Transport Layer 3-1

Chapter 3Transport Layer

Computer Networking A

Top Down Approach

6th edition Jim Kurose Keith

RossAddison-Wesley

March 2012

A note on the use of these ppt slidesWersquore making these slides freely available to all (faculty students readers) Theyrsquore in PowerPoint form so you see the animations and can add modify and delete slides (including this one) and slide content to suit your needs They obviously represent a lot of work on our part In return for use we only ask the following If you use these slides (eg in a class) that you mention their source

(after all wersquod like people to use our book) If you post any slides on a www site that you note that they are adapted

from (or perhaps identical to) our slides and note our copyright of this material

Thanks and enjoy JFKKWR

All material copyright 1996-2012 JF Kurose and KW Ross All Rights Reserved

Transport Layer 3-2

Chapter 3 Transport Layerour goals understand

principles behind transport layer services multiplexing

demultiplexing reliable data

transfer flow control congestion

control

learn about Internet transport layer protocols UDP connectionless

transport TCP connection-

oriented reliable transport

TCP congestion control

Transport Layer 3-3

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-4

Transport services and protocols

provide logical communication between app processes running on different hosts

transport protocols run in end systems send side breaks app

messages into segments passes to network layer

rcv side reassembles segments into messages passes to app layer

more than one transport protocol available to apps Internet TCP and UDP

application

transportnetworkdata linkphysical

logical end-end transport

application

transportnetworkdata linkphysical

Transport Layer 3-5

Transport vs network layer network layer

logical communication between hosts

transport layer logical communication between processes relies on

enhances network layer services

12 kids in Annrsquos house sending letters to 12 kids in Billrsquos house

hosts = houses processes = kids app messages =

letters in envelopes transport protocol =

Ann and Bill who demux to in-house siblings

network-layer protocol = postal service

household analogy

Transport Layer 3-6

Internet transport-layer protocols

reliable in-order delivery (TCP) congestion control flow control connection setup

unreliable unordered delivery UDP no-frills extension of

ldquobest-effortrdquo IP services not

available delay guarantees bandwidth

guarantees

application

transportnetworkdata linkphysical

application

transportnetworkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

logical end-end transport

Transport Layer 3-7

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-8

Multiplexingdemultiplexing

process

socket

use header info to deliverreceived segments to correct socket

demultiplexing at receiverhandle data from multiplesockets add transport header (later used for demultiplexing)

multiplexing at sender

transport

application

physical

link

network

P2P1

transport

application

physical

link

network

P4

transport

application

physical

link

network

P3

Transport Layer 3-9

How demultiplexing works

host receives IP datagrams each datagram has source IP

address destination IP address each datagram carries one

transport-layer segment each segment has source

destination port number host uses IP addresses amp port

numbers to direct segment to appropriate socket

source port dest port

32 bits

applicationdata

(payload)

other header fields

TCPUDP segment format

Transport Layer 3-10

Connectionless demultiplexing

recall created socket has host-local port

DatagramSocket mySocket1 = new DatagramSocket(12534)

when host receives UDP segment checks destination

port in segment directs UDP segment

to socket with that port

recall when creating datagram to send into UDP socket must specify

destination IP address destination port IP datagrams with same dest port but different source IP addresses andor source port numbers will be directed to same socket at dest

Transport Layer 3-11

Connectionless demux example

DatagramSocket serverSocket = new DatagramSocket

(6428)

transport

application

physical

link

network

P3transport

application

physical

link

network

P1

transport

application

physical

link

network

P4

DatagramSocket mySocket1 = new DatagramSocket (5775)

DatagramSocket mySocket2 = new DatagramSocket (9157)

source port 9157dest port 6428

source port 6428dest port 9157

source port dest port

source port dest port

Transport Layer 3-12

Connection-oriented demux

TCP socket identified by 4-tuple source IP address source port number dest IP address dest port number

demux receiver uses all four values to direct segment to appropriate socket

server host may support many simultaneous TCP sockets each socket identified

by its own 4-tuple web servers have

different sockets for each connecting client non-persistent HTTP

will have different socket for each request

Transport Layer 3-13

Connection-oriented demux example

transport

application

physical

link

network

P3transport

application

physical

link

P4

transport

application

physical

link

network

P2

source IPport A9157dest IP port B80

source IPport B80dest IPport A9157

host IP address

A

host IP address

C

network

P6P5P3

source IPport C5775dest IPport B80

source IPport C9157dest IPport B80

three segments all destined to IP address B dest port 80 are demultiplexed to different sockets

server IP

address B

Transport Layer 3-14

Connection-oriented demux example

transport

application

physical

link

network

P3transport

application

physical

link

transport

application

physical

link

network

P2

source IPport A9157dest IP port B80

source IPport B80dest IPport A9157

host IP address

A

host IP address

C

server IP

address B

network

P3

source IPport C5775dest IPport B80

source IPport C9157dest IPport B80

P4

threaded server

Transport Layer 3-15

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-16

UDP User Datagram Protocol [RFC 768]

ldquono frillsrdquo ldquobare bonesrdquo Internet transport protocol

ldquobest effortrdquo service UDP segments may be lost delivered out-of-order

to app connectionless

no handshaking between UDP sender receiver

each UDP segment handled independently of others

UDP use streaming

multimedia apps (loss tolerant rate sensitive)

DNS SNMP

reliable transfer over UDP add reliability at

application layer application-specific

error recovery

Transport Layer 3-17

UDP segment header

source port dest port

32 bits

applicationdata

(payload)

UDP segment format

length checksum

length in bytes of UDP segment

including header

no connection establishment (which can add delay)

simple no connection state at sender receiver

small header size no congestion control

UDP can blast away as fast as desired

why is there a UDP

Transport Layer 3-18

UDP checksum

sender treat segment contents

including header fields as sequence of 16-bit integers

checksum addition (onersquos complement sum) of segment contents

sender puts checksum value into UDP checksum field

receiver compute checksum of

received segment check if computed

checksum equals checksum field value NO - error detected YES - no error detected

But maybe errors nonetheless More later hellip

Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment

Transport Layer 3-19

Internet checksum exampleexample add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Note when adding numbers a carryout from the most significant bit needs to be added to the

result

Transport Layer 3-20

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-21

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

Transport Layer 3-23

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

important in application transport link layers top-10 list of important networking topics

Principles of reliable data transfer

Transport Layer 3-24

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-25

wersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt)

consider only unidirectional data transfer but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by

next event

eventactions

Reliable data transfer getting started

Transport Layer 3-26

rdt10 reliable transfer over a reliable channel underlying channel perfectly reliable

no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver reads data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-27

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

rdt20 channel with bit errors

How do humans recover from ldquoerrorsrdquoduring conversation

Transport Layer 3-28

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection feedback control msgs (ACKNAK) from

receiver to sender

rdt20 channel with bit errors

Transport Layer 3-29

rdt20 FSM specification

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

belowsender

receiverrdt_send(data)

Transport Layer 3-30

rdt20 operation with no errors

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-31

rdt20 error scenario

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-32

rdt20 has a fatal flaw

what happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

canrsquot just retransmit possible duplicate

handling duplicates sender retransmits current

pkt if ACKNAK corrupted sender adds sequence

number to each pkt receiver discards (doesnrsquot

deliver up) duplicate pkt

stop and waitsender sends one packet then waits for receiver response

Transport Layer 3-33

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-34

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt21 receiver handles garbled ACKNAKs

Transport Layer 3-35

rdt21 discussion

sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquoexpectedrdquo pkt should have seq of 0 or 1

receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-36

rdt22 a NAK-free protocol

same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last

pkt received OK receiver must explicitly include seq of pkt being

ACKed duplicate ACK at sender results in same

action as NAK retransmit current pkt

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-2

Chapter 3 Transport Layerour goals understand

principles behind transport layer services multiplexing

demultiplexing reliable data

transfer flow control congestion

control

learn about Internet transport layer protocols UDP connectionless

transport TCP connection-

oriented reliable transport

TCP congestion control

Transport Layer 3-3

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-4

Transport services and protocols

provide logical communication between app processes running on different hosts

transport protocols run in end systems send side breaks app

messages into segments passes to network layer

rcv side reassembles segments into messages passes to app layer

more than one transport protocol available to apps Internet TCP and UDP

application

transportnetworkdata linkphysical

logical end-end transport

application

transportnetworkdata linkphysical

Transport Layer 3-5

Transport vs network layer network layer

logical communication between hosts

transport layer logical communication between processes relies on

enhances network layer services

12 kids in Annrsquos house sending letters to 12 kids in Billrsquos house

hosts = houses processes = kids app messages =

letters in envelopes transport protocol =

Ann and Bill who demux to in-house siblings

network-layer protocol = postal service

household analogy

Transport Layer 3-6

Internet transport-layer protocols

reliable in-order delivery (TCP) congestion control flow control connection setup

unreliable unordered delivery UDP no-frills extension of

ldquobest-effortrdquo IP services not

available delay guarantees bandwidth

guarantees

application

transportnetworkdata linkphysical

application

transportnetworkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

logical end-end transport

Transport Layer 3-7

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-8

Multiplexingdemultiplexing

process

socket

use header info to deliverreceived segments to correct socket

demultiplexing at receiverhandle data from multiplesockets add transport header (later used for demultiplexing)

multiplexing at sender

transport

application

physical

link

network

P2P1

transport

application

physical

link

network

P4

transport

application

physical

link

network

P3

Transport Layer 3-9

How demultiplexing works

host receives IP datagrams each datagram has source IP

address destination IP address each datagram carries one

transport-layer segment each segment has source

destination port number host uses IP addresses amp port

numbers to direct segment to appropriate socket

source port dest port

32 bits

applicationdata

(payload)

other header fields

TCPUDP segment format

Transport Layer 3-10

Connectionless demultiplexing

recall created socket has host-local port

DatagramSocket mySocket1 = new DatagramSocket(12534)

when host receives UDP segment checks destination

port in segment directs UDP segment

to socket with that port

recall when creating datagram to send into UDP socket must specify

destination IP address destination port IP datagrams with same dest port but different source IP addresses andor source port numbers will be directed to same socket at dest

Transport Layer 3-11

Connectionless demux example

DatagramSocket serverSocket = new DatagramSocket

(6428)

transport

application

physical

link

network

P3transport

application

physical

link

network

P1

transport

application

physical

link

network

P4

DatagramSocket mySocket1 = new DatagramSocket (5775)

DatagramSocket mySocket2 = new DatagramSocket (9157)

source port 9157dest port 6428

source port 6428dest port 9157

source port dest port

source port dest port

Transport Layer 3-12

Connection-oriented demux

TCP socket identified by 4-tuple source IP address source port number dest IP address dest port number

demux receiver uses all four values to direct segment to appropriate socket

server host may support many simultaneous TCP sockets each socket identified

by its own 4-tuple web servers have

different sockets for each connecting client non-persistent HTTP

will have different socket for each request

Transport Layer 3-13

Connection-oriented demux example

transport

application

physical

link

network

P3transport

application

physical

link

P4

transport

application

physical

link

network

P2

source IPport A9157dest IP port B80

source IPport B80dest IPport A9157

host IP address

A

host IP address

C

network

P6P5P3

source IPport C5775dest IPport B80

source IPport C9157dest IPport B80

three segments all destined to IP address B dest port 80 are demultiplexed to different sockets

server IP

address B

Transport Layer 3-14

Connection-oriented demux example

transport

application

physical

link

network

P3transport

application

physical

link

transport

application

physical

link

network

P2

source IPport A9157dest IP port B80

source IPport B80dest IPport A9157

host IP address

A

host IP address

C

server IP

address B

network

P3

source IPport C5775dest IPport B80

source IPport C9157dest IPport B80

P4

threaded server

Transport Layer 3-15

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-16

UDP User Datagram Protocol [RFC 768]

ldquono frillsrdquo ldquobare bonesrdquo Internet transport protocol

ldquobest effortrdquo service UDP segments may be lost delivered out-of-order

to app connectionless

no handshaking between UDP sender receiver

each UDP segment handled independently of others

UDP use streaming

multimedia apps (loss tolerant rate sensitive)

DNS SNMP

reliable transfer over UDP add reliability at

application layer application-specific

error recovery

Transport Layer 3-17

UDP segment header

source port dest port

32 bits

applicationdata

(payload)

UDP segment format

length checksum

length in bytes of UDP segment

including header

no connection establishment (which can add delay)

simple no connection state at sender receiver

small header size no congestion control

UDP can blast away as fast as desired

why is there a UDP

Transport Layer 3-18

UDP checksum

sender treat segment contents

including header fields as sequence of 16-bit integers

checksum addition (onersquos complement sum) of segment contents

sender puts checksum value into UDP checksum field

receiver compute checksum of

received segment check if computed

checksum equals checksum field value NO - error detected YES - no error detected

But maybe errors nonetheless More later hellip

Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment

Transport Layer 3-19

Internet checksum exampleexample add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Note when adding numbers a carryout from the most significant bit needs to be added to the

result

Transport Layer 3-20

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-21

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

Transport Layer 3-23

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

important in application transport link layers top-10 list of important networking topics

Principles of reliable data transfer

Transport Layer 3-24

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-25

wersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt)

consider only unidirectional data transfer but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by

next event

eventactions

Reliable data transfer getting started

Transport Layer 3-26

rdt10 reliable transfer over a reliable channel underlying channel perfectly reliable

no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver reads data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-27

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

rdt20 channel with bit errors

How do humans recover from ldquoerrorsrdquoduring conversation

Transport Layer 3-28

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection feedback control msgs (ACKNAK) from

receiver to sender

rdt20 channel with bit errors

Transport Layer 3-29

rdt20 FSM specification

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

belowsender

receiverrdt_send(data)

Transport Layer 3-30

rdt20 operation with no errors

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-31

rdt20 error scenario

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-32

rdt20 has a fatal flaw

what happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

canrsquot just retransmit possible duplicate

handling duplicates sender retransmits current

pkt if ACKNAK corrupted sender adds sequence

number to each pkt receiver discards (doesnrsquot

deliver up) duplicate pkt

stop and waitsender sends one packet then waits for receiver response

Transport Layer 3-33

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-34

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt21 receiver handles garbled ACKNAKs

Transport Layer 3-35

rdt21 discussion

sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquoexpectedrdquo pkt should have seq of 0 or 1

receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-36

rdt22 a NAK-free protocol

same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last

pkt received OK receiver must explicitly include seq of pkt being

ACKed duplicate ACK at sender results in same

action as NAK retransmit current pkt

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-3

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-4

Transport services and protocols

provide logical communication between app processes running on different hosts

transport protocols run in end systems send side breaks app

messages into segments passes to network layer

rcv side reassembles segments into messages passes to app layer

more than one transport protocol available to apps Internet TCP and UDP

application

transportnetworkdata linkphysical

logical end-end transport

application

transportnetworkdata linkphysical

Transport Layer 3-5

Transport vs network layer network layer

logical communication between hosts

transport layer logical communication between processes relies on

enhances network layer services

12 kids in Annrsquos house sending letters to 12 kids in Billrsquos house

hosts = houses processes = kids app messages =

letters in envelopes transport protocol =

Ann and Bill who demux to in-house siblings

network-layer protocol = postal service

household analogy

Transport Layer 3-6

Internet transport-layer protocols

reliable in-order delivery (TCP) congestion control flow control connection setup

unreliable unordered delivery UDP no-frills extension of

ldquobest-effortrdquo IP services not

available delay guarantees bandwidth

guarantees

application

transportnetworkdata linkphysical

application

transportnetworkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

logical end-end transport

Transport Layer 3-7

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-8

Multiplexingdemultiplexing

process

socket

use header info to deliverreceived segments to correct socket

demultiplexing at receiverhandle data from multiplesockets add transport header (later used for demultiplexing)

multiplexing at sender

transport

application

physical

link

network

P2P1

transport

application

physical

link

network

P4

transport

application

physical

link

network

P3

Transport Layer 3-9

How demultiplexing works

host receives IP datagrams each datagram has source IP

address destination IP address each datagram carries one

transport-layer segment each segment has source

destination port number host uses IP addresses amp port

numbers to direct segment to appropriate socket

source port dest port

32 bits

applicationdata

(payload)

other header fields

TCPUDP segment format

Transport Layer 3-10

Connectionless demultiplexing

recall created socket has host-local port

DatagramSocket mySocket1 = new DatagramSocket(12534)

when host receives UDP segment checks destination

port in segment directs UDP segment

to socket with that port

recall when creating datagram to send into UDP socket must specify

destination IP address destination port IP datagrams with same dest port but different source IP addresses andor source port numbers will be directed to same socket at dest

Transport Layer 3-11

Connectionless demux example

DatagramSocket serverSocket = new DatagramSocket

(6428)

transport

application

physical

link

network

P3transport

application

physical

link

network

P1

transport

application

physical

link

network

P4

DatagramSocket mySocket1 = new DatagramSocket (5775)

DatagramSocket mySocket2 = new DatagramSocket (9157)

source port 9157dest port 6428

source port 6428dest port 9157

source port dest port

source port dest port

Transport Layer 3-12

Connection-oriented demux

TCP socket identified by 4-tuple source IP address source port number dest IP address dest port number

demux receiver uses all four values to direct segment to appropriate socket

server host may support many simultaneous TCP sockets each socket identified

by its own 4-tuple web servers have

different sockets for each connecting client non-persistent HTTP

will have different socket for each request

Transport Layer 3-13

Connection-oriented demux example

transport

application

physical

link

network

P3transport

application

physical

link

P4

transport

application

physical

link

network

P2

source IPport A9157dest IP port B80

source IPport B80dest IPport A9157

host IP address

A

host IP address

C

network

P6P5P3

source IPport C5775dest IPport B80

source IPport C9157dest IPport B80

three segments all destined to IP address B dest port 80 are demultiplexed to different sockets

server IP

address B

Transport Layer 3-14

Connection-oriented demux example

transport

application

physical

link

network

P3transport

application

physical

link

transport

application

physical

link

network

P2

source IPport A9157dest IP port B80

source IPport B80dest IPport A9157

host IP address

A

host IP address

C

server IP

address B

network

P3

source IPport C5775dest IPport B80

source IPport C9157dest IPport B80

P4

threaded server

Transport Layer 3-15

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-16

UDP User Datagram Protocol [RFC 768]

ldquono frillsrdquo ldquobare bonesrdquo Internet transport protocol

ldquobest effortrdquo service UDP segments may be lost delivered out-of-order

to app connectionless

no handshaking between UDP sender receiver

each UDP segment handled independently of others

UDP use streaming

multimedia apps (loss tolerant rate sensitive)

DNS SNMP

reliable transfer over UDP add reliability at

application layer application-specific

error recovery

Transport Layer 3-17

UDP segment header

source port dest port

32 bits

applicationdata

(payload)

UDP segment format

length checksum

length in bytes of UDP segment

including header

no connection establishment (which can add delay)

simple no connection state at sender receiver

small header size no congestion control

UDP can blast away as fast as desired

why is there a UDP

Transport Layer 3-18

UDP checksum

sender treat segment contents

including header fields as sequence of 16-bit integers

checksum addition (onersquos complement sum) of segment contents

sender puts checksum value into UDP checksum field

receiver compute checksum of

received segment check if computed

checksum equals checksum field value NO - error detected YES - no error detected

But maybe errors nonetheless More later hellip

Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment

Transport Layer 3-19

Internet checksum exampleexample add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Note when adding numbers a carryout from the most significant bit needs to be added to the

result

Transport Layer 3-20

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-21

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

Transport Layer 3-23

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

important in application transport link layers top-10 list of important networking topics

Principles of reliable data transfer

Transport Layer 3-24

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-25

wersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt)

consider only unidirectional data transfer but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by

next event

eventactions

Reliable data transfer getting started

Transport Layer 3-26

rdt10 reliable transfer over a reliable channel underlying channel perfectly reliable

no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver reads data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-27

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

rdt20 channel with bit errors

How do humans recover from ldquoerrorsrdquoduring conversation

Transport Layer 3-28

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection feedback control msgs (ACKNAK) from

receiver to sender

rdt20 channel with bit errors

Transport Layer 3-29

rdt20 FSM specification

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

belowsender

receiverrdt_send(data)

Transport Layer 3-30

rdt20 operation with no errors

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-31

rdt20 error scenario

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-32

rdt20 has a fatal flaw

what happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

canrsquot just retransmit possible duplicate

handling duplicates sender retransmits current

pkt if ACKNAK corrupted sender adds sequence

number to each pkt receiver discards (doesnrsquot

deliver up) duplicate pkt

stop and waitsender sends one packet then waits for receiver response

Transport Layer 3-33

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-34

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt21 receiver handles garbled ACKNAKs

Transport Layer 3-35

rdt21 discussion

sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquoexpectedrdquo pkt should have seq of 0 or 1

receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-36

rdt22 a NAK-free protocol

same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last

pkt received OK receiver must explicitly include seq of pkt being

ACKed duplicate ACK at sender results in same

action as NAK retransmit current pkt

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-4

Transport services and protocols

provide logical communication between app processes running on different hosts

transport protocols run in end systems send side breaks app

messages into segments passes to network layer

rcv side reassembles segments into messages passes to app layer

more than one transport protocol available to apps Internet TCP and UDP

application

transportnetworkdata linkphysical

logical end-end transport

application

transportnetworkdata linkphysical

Transport Layer 3-5

Transport vs network layer network layer

logical communication between hosts

transport layer logical communication between processes relies on

enhances network layer services

12 kids in Annrsquos house sending letters to 12 kids in Billrsquos house

hosts = houses processes = kids app messages =

letters in envelopes transport protocol =

Ann and Bill who demux to in-house siblings

network-layer protocol = postal service

household analogy

Transport Layer 3-6

Internet transport-layer protocols

reliable in-order delivery (TCP) congestion control flow control connection setup

unreliable unordered delivery UDP no-frills extension of

ldquobest-effortrdquo IP services not

available delay guarantees bandwidth

guarantees

application

transportnetworkdata linkphysical

application

transportnetworkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

logical end-end transport

Transport Layer 3-7

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-8

Multiplexingdemultiplexing

process

socket

use header info to deliverreceived segments to correct socket

demultiplexing at receiverhandle data from multiplesockets add transport header (later used for demultiplexing)

multiplexing at sender

transport

application

physical

link

network

P2P1

transport

application

physical

link

network

P4

transport

application

physical

link

network

P3

Transport Layer 3-9

How demultiplexing works

host receives IP datagrams each datagram has source IP

address destination IP address each datagram carries one

transport-layer segment each segment has source

destination port number host uses IP addresses amp port

numbers to direct segment to appropriate socket

source port dest port

32 bits

applicationdata

(payload)

other header fields

TCPUDP segment format

Transport Layer 3-10

Connectionless demultiplexing

recall created socket has host-local port

DatagramSocket mySocket1 = new DatagramSocket(12534)

when host receives UDP segment checks destination

port in segment directs UDP segment

to socket with that port

recall when creating datagram to send into UDP socket must specify

destination IP address destination port IP datagrams with same dest port but different source IP addresses andor source port numbers will be directed to same socket at dest

Transport Layer 3-11

Connectionless demux example

DatagramSocket serverSocket = new DatagramSocket

(6428)

transport

application

physical

link

network

P3transport

application

physical

link

network

P1

transport

application

physical

link

network

P4

DatagramSocket mySocket1 = new DatagramSocket (5775)

DatagramSocket mySocket2 = new DatagramSocket (9157)

source port 9157dest port 6428

source port 6428dest port 9157

source port dest port

source port dest port

Transport Layer 3-12

Connection-oriented demux

TCP socket identified by 4-tuple source IP address source port number dest IP address dest port number

demux receiver uses all four values to direct segment to appropriate socket

server host may support many simultaneous TCP sockets each socket identified

by its own 4-tuple web servers have

different sockets for each connecting client non-persistent HTTP

will have different socket for each request

Transport Layer 3-13

Connection-oriented demux example

transport

application

physical

link

network

P3transport

application

physical

link

P4

transport

application

physical

link

network

P2

source IPport A9157dest IP port B80

source IPport B80dest IPport A9157

host IP address

A

host IP address

C

network

P6P5P3

source IPport C5775dest IPport B80

source IPport C9157dest IPport B80

three segments all destined to IP address B dest port 80 are demultiplexed to different sockets

server IP

address B

Transport Layer 3-14

Connection-oriented demux example

transport

application

physical

link

network

P3transport

application

physical

link

transport

application

physical

link

network

P2

source IPport A9157dest IP port B80

source IPport B80dest IPport A9157

host IP address

A

host IP address

C

server IP

address B

network

P3

source IPport C5775dest IPport B80

source IPport C9157dest IPport B80

P4

threaded server

Transport Layer 3-15

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-16

UDP User Datagram Protocol [RFC 768]

ldquono frillsrdquo ldquobare bonesrdquo Internet transport protocol

ldquobest effortrdquo service UDP segments may be lost delivered out-of-order

to app connectionless

no handshaking between UDP sender receiver

each UDP segment handled independently of others

UDP use streaming

multimedia apps (loss tolerant rate sensitive)

DNS SNMP

reliable transfer over UDP add reliability at

application layer application-specific

error recovery

Transport Layer 3-17

UDP segment header

source port dest port

32 bits

applicationdata

(payload)

UDP segment format

length checksum

length in bytes of UDP segment

including header

no connection establishment (which can add delay)

simple no connection state at sender receiver

small header size no congestion control

UDP can blast away as fast as desired

why is there a UDP

Transport Layer 3-18

UDP checksum

sender treat segment contents

including header fields as sequence of 16-bit integers

checksum addition (onersquos complement sum) of segment contents

sender puts checksum value into UDP checksum field

receiver compute checksum of

received segment check if computed

checksum equals checksum field value NO - error detected YES - no error detected

But maybe errors nonetheless More later hellip

Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment

Transport Layer 3-19

Internet checksum exampleexample add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Note when adding numbers a carryout from the most significant bit needs to be added to the

result

Transport Layer 3-20

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-21

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

Transport Layer 3-23

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

important in application transport link layers top-10 list of important networking topics

Principles of reliable data transfer

Transport Layer 3-24

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-25

wersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt)

consider only unidirectional data transfer but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by

next event

eventactions

Reliable data transfer getting started

Transport Layer 3-26

rdt10 reliable transfer over a reliable channel underlying channel perfectly reliable

no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver reads data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-27

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

rdt20 channel with bit errors

How do humans recover from ldquoerrorsrdquoduring conversation

Transport Layer 3-28

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection feedback control msgs (ACKNAK) from

receiver to sender

rdt20 channel with bit errors

Transport Layer 3-29

rdt20 FSM specification

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

belowsender

receiverrdt_send(data)

Transport Layer 3-30

rdt20 operation with no errors

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-31

rdt20 error scenario

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-32

rdt20 has a fatal flaw

what happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

canrsquot just retransmit possible duplicate

handling duplicates sender retransmits current

pkt if ACKNAK corrupted sender adds sequence

number to each pkt receiver discards (doesnrsquot

deliver up) duplicate pkt

stop and waitsender sends one packet then waits for receiver response

Transport Layer 3-33

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-34

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt21 receiver handles garbled ACKNAKs

Transport Layer 3-35

rdt21 discussion

sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquoexpectedrdquo pkt should have seq of 0 or 1

receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-36

rdt22 a NAK-free protocol

same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last

pkt received OK receiver must explicitly include seq of pkt being

ACKed duplicate ACK at sender results in same

action as NAK retransmit current pkt

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-5

Transport vs network layer network layer

logical communication between hosts

transport layer logical communication between processes relies on

enhances network layer services

12 kids in Annrsquos house sending letters to 12 kids in Billrsquos house

hosts = houses processes = kids app messages =

letters in envelopes transport protocol =

Ann and Bill who demux to in-house siblings

network-layer protocol = postal service

household analogy

Transport Layer 3-6

Internet transport-layer protocols

reliable in-order delivery (TCP) congestion control flow control connection setup

unreliable unordered delivery UDP no-frills extension of

ldquobest-effortrdquo IP services not

available delay guarantees bandwidth

guarantees

application

transportnetworkdata linkphysical

application

transportnetworkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

logical end-end transport

Transport Layer 3-7

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-8

Multiplexingdemultiplexing

process

socket

use header info to deliverreceived segments to correct socket

demultiplexing at receiverhandle data from multiplesockets add transport header (later used for demultiplexing)

multiplexing at sender

transport

application

physical

link

network

P2P1

transport

application

physical

link

network

P4

transport

application

physical

link

network

P3

Transport Layer 3-9

How demultiplexing works

host receives IP datagrams each datagram has source IP

address destination IP address each datagram carries one

transport-layer segment each segment has source

destination port number host uses IP addresses amp port

numbers to direct segment to appropriate socket

source port dest port

32 bits

applicationdata

(payload)

other header fields

TCPUDP segment format

Transport Layer 3-10

Connectionless demultiplexing

recall created socket has host-local port

DatagramSocket mySocket1 = new DatagramSocket(12534)

when host receives UDP segment checks destination

port in segment directs UDP segment

to socket with that port

recall when creating datagram to send into UDP socket must specify

destination IP address destination port IP datagrams with same dest port but different source IP addresses andor source port numbers will be directed to same socket at dest

Transport Layer 3-11

Connectionless demux example

DatagramSocket serverSocket = new DatagramSocket

(6428)

transport

application

physical

link

network

P3transport

application

physical

link

network

P1

transport

application

physical

link

network

P4

DatagramSocket mySocket1 = new DatagramSocket (5775)

DatagramSocket mySocket2 = new DatagramSocket (9157)

source port 9157dest port 6428

source port 6428dest port 9157

source port dest port

source port dest port

Transport Layer 3-12

Connection-oriented demux

TCP socket identified by 4-tuple source IP address source port number dest IP address dest port number

demux receiver uses all four values to direct segment to appropriate socket

server host may support many simultaneous TCP sockets each socket identified

by its own 4-tuple web servers have

different sockets for each connecting client non-persistent HTTP

will have different socket for each request

Transport Layer 3-13

Connection-oriented demux example

transport

application

physical

link

network

P3transport

application

physical

link

P4

transport

application

physical

link

network

P2

source IPport A9157dest IP port B80

source IPport B80dest IPport A9157

host IP address

A

host IP address

C

network

P6P5P3

source IPport C5775dest IPport B80

source IPport C9157dest IPport B80

three segments all destined to IP address B dest port 80 are demultiplexed to different sockets

server IP

address B

Transport Layer 3-14

Connection-oriented demux example

transport

application

physical

link

network

P3transport

application

physical

link

transport

application

physical

link

network

P2

source IPport A9157dest IP port B80

source IPport B80dest IPport A9157

host IP address

A

host IP address

C

server IP

address B

network

P3

source IPport C5775dest IPport B80

source IPport C9157dest IPport B80

P4

threaded server

Transport Layer 3-15

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-16

UDP User Datagram Protocol [RFC 768]

ldquono frillsrdquo ldquobare bonesrdquo Internet transport protocol

ldquobest effortrdquo service UDP segments may be lost delivered out-of-order

to app connectionless

no handshaking between UDP sender receiver

each UDP segment handled independently of others

UDP use streaming

multimedia apps (loss tolerant rate sensitive)

DNS SNMP

reliable transfer over UDP add reliability at

application layer application-specific

error recovery

Transport Layer 3-17

UDP segment header

source port dest port

32 bits

applicationdata

(payload)

UDP segment format

length checksum

length in bytes of UDP segment

including header

no connection establishment (which can add delay)

simple no connection state at sender receiver

small header size no congestion control

UDP can blast away as fast as desired

why is there a UDP

Transport Layer 3-18

UDP checksum

sender treat segment contents

including header fields as sequence of 16-bit integers

checksum addition (onersquos complement sum) of segment contents

sender puts checksum value into UDP checksum field

receiver compute checksum of

received segment check if computed

checksum equals checksum field value NO - error detected YES - no error detected

But maybe errors nonetheless More later hellip

Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment

Transport Layer 3-19

Internet checksum exampleexample add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Note when adding numbers a carryout from the most significant bit needs to be added to the

result

Transport Layer 3-20

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-21

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

Transport Layer 3-23

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

important in application transport link layers top-10 list of important networking topics

Principles of reliable data transfer

Transport Layer 3-24

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-25

wersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt)

consider only unidirectional data transfer but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by

next event

eventactions

Reliable data transfer getting started

Transport Layer 3-26

rdt10 reliable transfer over a reliable channel underlying channel perfectly reliable

no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver reads data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-27

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

rdt20 channel with bit errors

How do humans recover from ldquoerrorsrdquoduring conversation

Transport Layer 3-28

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection feedback control msgs (ACKNAK) from

receiver to sender

rdt20 channel with bit errors

Transport Layer 3-29

rdt20 FSM specification

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

belowsender

receiverrdt_send(data)

Transport Layer 3-30

rdt20 operation with no errors

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-31

rdt20 error scenario

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-32

rdt20 has a fatal flaw

what happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

canrsquot just retransmit possible duplicate

handling duplicates sender retransmits current

pkt if ACKNAK corrupted sender adds sequence

number to each pkt receiver discards (doesnrsquot

deliver up) duplicate pkt

stop and waitsender sends one packet then waits for receiver response

Transport Layer 3-33

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-34

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt21 receiver handles garbled ACKNAKs

Transport Layer 3-35

rdt21 discussion

sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquoexpectedrdquo pkt should have seq of 0 or 1

receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-36

rdt22 a NAK-free protocol

same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last

pkt received OK receiver must explicitly include seq of pkt being

ACKed duplicate ACK at sender results in same

action as NAK retransmit current pkt

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-6

Internet transport-layer protocols

reliable in-order delivery (TCP) congestion control flow control connection setup

unreliable unordered delivery UDP no-frills extension of

ldquobest-effortrdquo IP services not

available delay guarantees bandwidth

guarantees

application

transportnetworkdata linkphysical

application

transportnetworkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

logical end-end transport

Transport Layer 3-7

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-8

Multiplexingdemultiplexing

process

socket

use header info to deliverreceived segments to correct socket

demultiplexing at receiverhandle data from multiplesockets add transport header (later used for demultiplexing)

multiplexing at sender

transport

application

physical

link

network

P2P1

transport

application

physical

link

network

P4

transport

application

physical

link

network

P3

Transport Layer 3-9

How demultiplexing works

host receives IP datagrams each datagram has source IP

address destination IP address each datagram carries one

transport-layer segment each segment has source

destination port number host uses IP addresses amp port

numbers to direct segment to appropriate socket

source port dest port

32 bits

applicationdata

(payload)

other header fields

TCPUDP segment format

Transport Layer 3-10

Connectionless demultiplexing

recall created socket has host-local port

DatagramSocket mySocket1 = new DatagramSocket(12534)

when host receives UDP segment checks destination

port in segment directs UDP segment

to socket with that port

recall when creating datagram to send into UDP socket must specify

destination IP address destination port IP datagrams with same dest port but different source IP addresses andor source port numbers will be directed to same socket at dest

Transport Layer 3-11

Connectionless demux example

DatagramSocket serverSocket = new DatagramSocket

(6428)

transport

application

physical

link

network

P3transport

application

physical

link

network

P1

transport

application

physical

link

network

P4

DatagramSocket mySocket1 = new DatagramSocket (5775)

DatagramSocket mySocket2 = new DatagramSocket (9157)

source port 9157dest port 6428

source port 6428dest port 9157

source port dest port

source port dest port

Transport Layer 3-12

Connection-oriented demux

TCP socket identified by 4-tuple source IP address source port number dest IP address dest port number

demux receiver uses all four values to direct segment to appropriate socket

server host may support many simultaneous TCP sockets each socket identified

by its own 4-tuple web servers have

different sockets for each connecting client non-persistent HTTP

will have different socket for each request

Transport Layer 3-13

Connection-oriented demux example

transport

application

physical

link

network

P3transport

application

physical

link

P4

transport

application

physical

link

network

P2

source IPport A9157dest IP port B80

source IPport B80dest IPport A9157

host IP address

A

host IP address

C

network

P6P5P3

source IPport C5775dest IPport B80

source IPport C9157dest IPport B80

three segments all destined to IP address B dest port 80 are demultiplexed to different sockets

server IP

address B

Transport Layer 3-14

Connection-oriented demux example

transport

application

physical

link

network

P3transport

application

physical

link

transport

application

physical

link

network

P2

source IPport A9157dest IP port B80

source IPport B80dest IPport A9157

host IP address

A

host IP address

C

server IP

address B

network

P3

source IPport C5775dest IPport B80

source IPport C9157dest IPport B80

P4

threaded server

Transport Layer 3-15

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-16

UDP User Datagram Protocol [RFC 768]

ldquono frillsrdquo ldquobare bonesrdquo Internet transport protocol

ldquobest effortrdquo service UDP segments may be lost delivered out-of-order

to app connectionless

no handshaking between UDP sender receiver

each UDP segment handled independently of others

UDP use streaming

multimedia apps (loss tolerant rate sensitive)

DNS SNMP

reliable transfer over UDP add reliability at

application layer application-specific

error recovery

Transport Layer 3-17

UDP segment header

source port dest port

32 bits

applicationdata

(payload)

UDP segment format

length checksum

length in bytes of UDP segment

including header

no connection establishment (which can add delay)

simple no connection state at sender receiver

small header size no congestion control

UDP can blast away as fast as desired

why is there a UDP

Transport Layer 3-18

UDP checksum

sender treat segment contents

including header fields as sequence of 16-bit integers

checksum addition (onersquos complement sum) of segment contents

sender puts checksum value into UDP checksum field

receiver compute checksum of

received segment check if computed

checksum equals checksum field value NO - error detected YES - no error detected

But maybe errors nonetheless More later hellip

Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment

Transport Layer 3-19

Internet checksum exampleexample add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Note when adding numbers a carryout from the most significant bit needs to be added to the

result

Transport Layer 3-20

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-21

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

Transport Layer 3-23

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

important in application transport link layers top-10 list of important networking topics

Principles of reliable data transfer

Transport Layer 3-24

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-25

wersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt)

consider only unidirectional data transfer but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by

next event

eventactions

Reliable data transfer getting started

Transport Layer 3-26

rdt10 reliable transfer over a reliable channel underlying channel perfectly reliable

no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver reads data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-27

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

rdt20 channel with bit errors

How do humans recover from ldquoerrorsrdquoduring conversation

Transport Layer 3-28

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection feedback control msgs (ACKNAK) from

receiver to sender

rdt20 channel with bit errors

Transport Layer 3-29

rdt20 FSM specification

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

belowsender

receiverrdt_send(data)

Transport Layer 3-30

rdt20 operation with no errors

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-31

rdt20 error scenario

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-32

rdt20 has a fatal flaw

what happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

canrsquot just retransmit possible duplicate

handling duplicates sender retransmits current

pkt if ACKNAK corrupted sender adds sequence

number to each pkt receiver discards (doesnrsquot

deliver up) duplicate pkt

stop and waitsender sends one packet then waits for receiver response

Transport Layer 3-33

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-34

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt21 receiver handles garbled ACKNAKs

Transport Layer 3-35

rdt21 discussion

sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquoexpectedrdquo pkt should have seq of 0 or 1

receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-36

rdt22 a NAK-free protocol

same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last

pkt received OK receiver must explicitly include seq of pkt being

ACKed duplicate ACK at sender results in same

action as NAK retransmit current pkt

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-7

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-8

Multiplexingdemultiplexing

process

socket

use header info to deliverreceived segments to correct socket

demultiplexing at receiverhandle data from multiplesockets add transport header (later used for demultiplexing)

multiplexing at sender

transport

application

physical

link

network

P2P1

transport

application

physical

link

network

P4

transport

application

physical

link

network

P3

Transport Layer 3-9

How demultiplexing works

host receives IP datagrams each datagram has source IP

address destination IP address each datagram carries one

transport-layer segment each segment has source

destination port number host uses IP addresses amp port

numbers to direct segment to appropriate socket

source port dest port

32 bits

applicationdata

(payload)

other header fields

TCPUDP segment format

Transport Layer 3-10

Connectionless demultiplexing

recall created socket has host-local port

DatagramSocket mySocket1 = new DatagramSocket(12534)

when host receives UDP segment checks destination

port in segment directs UDP segment

to socket with that port

recall when creating datagram to send into UDP socket must specify

destination IP address destination port IP datagrams with same dest port but different source IP addresses andor source port numbers will be directed to same socket at dest

Transport Layer 3-11

Connectionless demux example

DatagramSocket serverSocket = new DatagramSocket

(6428)

transport

application

physical

link

network

P3transport

application

physical

link

network

P1

transport

application

physical

link

network

P4

DatagramSocket mySocket1 = new DatagramSocket (5775)

DatagramSocket mySocket2 = new DatagramSocket (9157)

source port 9157dest port 6428

source port 6428dest port 9157

source port dest port

source port dest port

Transport Layer 3-12

Connection-oriented demux

TCP socket identified by 4-tuple source IP address source port number dest IP address dest port number

demux receiver uses all four values to direct segment to appropriate socket

server host may support many simultaneous TCP sockets each socket identified

by its own 4-tuple web servers have

different sockets for each connecting client non-persistent HTTP

will have different socket for each request

Transport Layer 3-13

Connection-oriented demux example

transport

application

physical

link

network

P3transport

application

physical

link

P4

transport

application

physical

link

network

P2

source IPport A9157dest IP port B80

source IPport B80dest IPport A9157

host IP address

A

host IP address

C

network

P6P5P3

source IPport C5775dest IPport B80

source IPport C9157dest IPport B80

three segments all destined to IP address B dest port 80 are demultiplexed to different sockets

server IP

address B

Transport Layer 3-14

Connection-oriented demux example

transport

application

physical

link

network

P3transport

application

physical

link

transport

application

physical

link

network

P2

source IPport A9157dest IP port B80

source IPport B80dest IPport A9157

host IP address

A

host IP address

C

server IP

address B

network

P3

source IPport C5775dest IPport B80

source IPport C9157dest IPport B80

P4

threaded server

Transport Layer 3-15

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-16

UDP User Datagram Protocol [RFC 768]

ldquono frillsrdquo ldquobare bonesrdquo Internet transport protocol

ldquobest effortrdquo service UDP segments may be lost delivered out-of-order

to app connectionless

no handshaking between UDP sender receiver

each UDP segment handled independently of others

UDP use streaming

multimedia apps (loss tolerant rate sensitive)

DNS SNMP

reliable transfer over UDP add reliability at

application layer application-specific

error recovery

Transport Layer 3-17

UDP segment header

source port dest port

32 bits

applicationdata

(payload)

UDP segment format

length checksum

length in bytes of UDP segment

including header

no connection establishment (which can add delay)

simple no connection state at sender receiver

small header size no congestion control

UDP can blast away as fast as desired

why is there a UDP

Transport Layer 3-18

UDP checksum

sender treat segment contents

including header fields as sequence of 16-bit integers

checksum addition (onersquos complement sum) of segment contents

sender puts checksum value into UDP checksum field

receiver compute checksum of

received segment check if computed

checksum equals checksum field value NO - error detected YES - no error detected

But maybe errors nonetheless More later hellip

Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment

Transport Layer 3-19

Internet checksum exampleexample add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Note when adding numbers a carryout from the most significant bit needs to be added to the

result

Transport Layer 3-20

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-21

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

Transport Layer 3-23

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

important in application transport link layers top-10 list of important networking topics

Principles of reliable data transfer

Transport Layer 3-24

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-25

wersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt)

consider only unidirectional data transfer but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by

next event

eventactions

Reliable data transfer getting started

Transport Layer 3-26

rdt10 reliable transfer over a reliable channel underlying channel perfectly reliable

no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver reads data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-27

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

rdt20 channel with bit errors

How do humans recover from ldquoerrorsrdquoduring conversation

Transport Layer 3-28

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection feedback control msgs (ACKNAK) from

receiver to sender

rdt20 channel with bit errors

Transport Layer 3-29

rdt20 FSM specification

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

belowsender

receiverrdt_send(data)

Transport Layer 3-30

rdt20 operation with no errors

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-31

rdt20 error scenario

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-32

rdt20 has a fatal flaw

what happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

canrsquot just retransmit possible duplicate

handling duplicates sender retransmits current

pkt if ACKNAK corrupted sender adds sequence

number to each pkt receiver discards (doesnrsquot

deliver up) duplicate pkt

stop and waitsender sends one packet then waits for receiver response

Transport Layer 3-33

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-34

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt21 receiver handles garbled ACKNAKs

Transport Layer 3-35

rdt21 discussion

sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquoexpectedrdquo pkt should have seq of 0 or 1

receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-36

rdt22 a NAK-free protocol

same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last

pkt received OK receiver must explicitly include seq of pkt being

ACKed duplicate ACK at sender results in same

action as NAK retransmit current pkt

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-8

Multiplexingdemultiplexing

process

socket

use header info to deliverreceived segments to correct socket

demultiplexing at receiverhandle data from multiplesockets add transport header (later used for demultiplexing)

multiplexing at sender

transport

application

physical

link

network

P2P1

transport

application

physical

link

network

P4

transport

application

physical

link

network

P3

Transport Layer 3-9

How demultiplexing works

host receives IP datagrams each datagram has source IP

address destination IP address each datagram carries one

transport-layer segment each segment has source

destination port number host uses IP addresses amp port

numbers to direct segment to appropriate socket

source port dest port

32 bits

applicationdata

(payload)

other header fields

TCPUDP segment format

Transport Layer 3-10

Connectionless demultiplexing

recall created socket has host-local port

DatagramSocket mySocket1 = new DatagramSocket(12534)

when host receives UDP segment checks destination

port in segment directs UDP segment

to socket with that port

recall when creating datagram to send into UDP socket must specify

destination IP address destination port IP datagrams with same dest port but different source IP addresses andor source port numbers will be directed to same socket at dest

Transport Layer 3-11

Connectionless demux example

DatagramSocket serverSocket = new DatagramSocket

(6428)

transport

application

physical

link

network

P3transport

application

physical

link

network

P1

transport

application

physical

link

network

P4

DatagramSocket mySocket1 = new DatagramSocket (5775)

DatagramSocket mySocket2 = new DatagramSocket (9157)

source port 9157dest port 6428

source port 6428dest port 9157

source port dest port

source port dest port

Transport Layer 3-12

Connection-oriented demux

TCP socket identified by 4-tuple source IP address source port number dest IP address dest port number

demux receiver uses all four values to direct segment to appropriate socket

server host may support many simultaneous TCP sockets each socket identified

by its own 4-tuple web servers have

different sockets for each connecting client non-persistent HTTP

will have different socket for each request

Transport Layer 3-13

Connection-oriented demux example

transport

application

physical

link

network

P3transport

application

physical

link

P4

transport

application

physical

link

network

P2

source IPport A9157dest IP port B80

source IPport B80dest IPport A9157

host IP address

A

host IP address

C

network

P6P5P3

source IPport C5775dest IPport B80

source IPport C9157dest IPport B80

three segments all destined to IP address B dest port 80 are demultiplexed to different sockets

server IP

address B

Transport Layer 3-14

Connection-oriented demux example

transport

application

physical

link

network

P3transport

application

physical

link

transport

application

physical

link

network

P2

source IPport A9157dest IP port B80

source IPport B80dest IPport A9157

host IP address

A

host IP address

C

server IP

address B

network

P3

source IPport C5775dest IPport B80

source IPport C9157dest IPport B80

P4

threaded server

Transport Layer 3-15

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-16

UDP User Datagram Protocol [RFC 768]

ldquono frillsrdquo ldquobare bonesrdquo Internet transport protocol

ldquobest effortrdquo service UDP segments may be lost delivered out-of-order

to app connectionless

no handshaking between UDP sender receiver

each UDP segment handled independently of others

UDP use streaming

multimedia apps (loss tolerant rate sensitive)

DNS SNMP

reliable transfer over UDP add reliability at

application layer application-specific

error recovery

Transport Layer 3-17

UDP segment header

source port dest port

32 bits

applicationdata

(payload)

UDP segment format

length checksum

length in bytes of UDP segment

including header

no connection establishment (which can add delay)

simple no connection state at sender receiver

small header size no congestion control

UDP can blast away as fast as desired

why is there a UDP

Transport Layer 3-18

UDP checksum

sender treat segment contents

including header fields as sequence of 16-bit integers

checksum addition (onersquos complement sum) of segment contents

sender puts checksum value into UDP checksum field

receiver compute checksum of

received segment check if computed

checksum equals checksum field value NO - error detected YES - no error detected

But maybe errors nonetheless More later hellip

Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment

Transport Layer 3-19

Internet checksum exampleexample add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Note when adding numbers a carryout from the most significant bit needs to be added to the

result

Transport Layer 3-20

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-21

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

Transport Layer 3-23

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

important in application transport link layers top-10 list of important networking topics

Principles of reliable data transfer

Transport Layer 3-24

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-25

wersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt)

consider only unidirectional data transfer but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by

next event

eventactions

Reliable data transfer getting started

Transport Layer 3-26

rdt10 reliable transfer over a reliable channel underlying channel perfectly reliable

no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver reads data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-27

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

rdt20 channel with bit errors

How do humans recover from ldquoerrorsrdquoduring conversation

Transport Layer 3-28

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection feedback control msgs (ACKNAK) from

receiver to sender

rdt20 channel with bit errors

Transport Layer 3-29

rdt20 FSM specification

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

belowsender

receiverrdt_send(data)

Transport Layer 3-30

rdt20 operation with no errors

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-31

rdt20 error scenario

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-32

rdt20 has a fatal flaw

what happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

canrsquot just retransmit possible duplicate

handling duplicates sender retransmits current

pkt if ACKNAK corrupted sender adds sequence

number to each pkt receiver discards (doesnrsquot

deliver up) duplicate pkt

stop and waitsender sends one packet then waits for receiver response

Transport Layer 3-33

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-34

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt21 receiver handles garbled ACKNAKs

Transport Layer 3-35

rdt21 discussion

sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquoexpectedrdquo pkt should have seq of 0 or 1

receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-36

rdt22 a NAK-free protocol

same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last

pkt received OK receiver must explicitly include seq of pkt being

ACKed duplicate ACK at sender results in same

action as NAK retransmit current pkt

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-9

How demultiplexing works

host receives IP datagrams each datagram has source IP

address destination IP address each datagram carries one

transport-layer segment each segment has source

destination port number host uses IP addresses amp port

numbers to direct segment to appropriate socket

source port dest port

32 bits

applicationdata

(payload)

other header fields

TCPUDP segment format

Transport Layer 3-10

Connectionless demultiplexing

recall created socket has host-local port

DatagramSocket mySocket1 = new DatagramSocket(12534)

when host receives UDP segment checks destination

port in segment directs UDP segment

to socket with that port

recall when creating datagram to send into UDP socket must specify

destination IP address destination port IP datagrams with same dest port but different source IP addresses andor source port numbers will be directed to same socket at dest

Transport Layer 3-11

Connectionless demux example

DatagramSocket serverSocket = new DatagramSocket

(6428)

transport

application

physical

link

network

P3transport

application

physical

link

network

P1

transport

application

physical

link

network

P4

DatagramSocket mySocket1 = new DatagramSocket (5775)

DatagramSocket mySocket2 = new DatagramSocket (9157)

source port 9157dest port 6428

source port 6428dest port 9157

source port dest port

source port dest port

Transport Layer 3-12

Connection-oriented demux

TCP socket identified by 4-tuple source IP address source port number dest IP address dest port number

demux receiver uses all four values to direct segment to appropriate socket

server host may support many simultaneous TCP sockets each socket identified

by its own 4-tuple web servers have

different sockets for each connecting client non-persistent HTTP

will have different socket for each request

Transport Layer 3-13

Connection-oriented demux example

transport

application

physical

link

network

P3transport

application

physical

link

P4

transport

application

physical

link

network

P2

source IPport A9157dest IP port B80

source IPport B80dest IPport A9157

host IP address

A

host IP address

C

network

P6P5P3

source IPport C5775dest IPport B80

source IPport C9157dest IPport B80

three segments all destined to IP address B dest port 80 are demultiplexed to different sockets

server IP

address B

Transport Layer 3-14

Connection-oriented demux example

transport

application

physical

link

network

P3transport

application

physical

link

transport

application

physical

link

network

P2

source IPport A9157dest IP port B80

source IPport B80dest IPport A9157

host IP address

A

host IP address

C

server IP

address B

network

P3

source IPport C5775dest IPport B80

source IPport C9157dest IPport B80

P4

threaded server

Transport Layer 3-15

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-16

UDP User Datagram Protocol [RFC 768]

ldquono frillsrdquo ldquobare bonesrdquo Internet transport protocol

ldquobest effortrdquo service UDP segments may be lost delivered out-of-order

to app connectionless

no handshaking between UDP sender receiver

each UDP segment handled independently of others

UDP use streaming

multimedia apps (loss tolerant rate sensitive)

DNS SNMP

reliable transfer over UDP add reliability at

application layer application-specific

error recovery

Transport Layer 3-17

UDP segment header

source port dest port

32 bits

applicationdata

(payload)

UDP segment format

length checksum

length in bytes of UDP segment

including header

no connection establishment (which can add delay)

simple no connection state at sender receiver

small header size no congestion control

UDP can blast away as fast as desired

why is there a UDP

Transport Layer 3-18

UDP checksum

sender treat segment contents

including header fields as sequence of 16-bit integers

checksum addition (onersquos complement sum) of segment contents

sender puts checksum value into UDP checksum field

receiver compute checksum of

received segment check if computed

checksum equals checksum field value NO - error detected YES - no error detected

But maybe errors nonetheless More later hellip

Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment

Transport Layer 3-19

Internet checksum exampleexample add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Note when adding numbers a carryout from the most significant bit needs to be added to the

result

Transport Layer 3-20

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-21

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

Transport Layer 3-23

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

important in application transport link layers top-10 list of important networking topics

Principles of reliable data transfer

Transport Layer 3-24

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-25

wersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt)

consider only unidirectional data transfer but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by

next event

eventactions

Reliable data transfer getting started

Transport Layer 3-26

rdt10 reliable transfer over a reliable channel underlying channel perfectly reliable

no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver reads data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-27

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

rdt20 channel with bit errors

How do humans recover from ldquoerrorsrdquoduring conversation

Transport Layer 3-28

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection feedback control msgs (ACKNAK) from

receiver to sender

rdt20 channel with bit errors

Transport Layer 3-29

rdt20 FSM specification

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

belowsender

receiverrdt_send(data)

Transport Layer 3-30

rdt20 operation with no errors

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-31

rdt20 error scenario

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-32

rdt20 has a fatal flaw

what happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

canrsquot just retransmit possible duplicate

handling duplicates sender retransmits current

pkt if ACKNAK corrupted sender adds sequence

number to each pkt receiver discards (doesnrsquot

deliver up) duplicate pkt

stop and waitsender sends one packet then waits for receiver response

Transport Layer 3-33

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-34

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt21 receiver handles garbled ACKNAKs

Transport Layer 3-35

rdt21 discussion

sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquoexpectedrdquo pkt should have seq of 0 or 1

receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-36

rdt22 a NAK-free protocol

same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last

pkt received OK receiver must explicitly include seq of pkt being

ACKed duplicate ACK at sender results in same

action as NAK retransmit current pkt

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-10

Connectionless demultiplexing

recall created socket has host-local port

DatagramSocket mySocket1 = new DatagramSocket(12534)

when host receives UDP segment checks destination

port in segment directs UDP segment

to socket with that port

recall when creating datagram to send into UDP socket must specify

destination IP address destination port IP datagrams with same dest port but different source IP addresses andor source port numbers will be directed to same socket at dest

Transport Layer 3-11

Connectionless demux example

DatagramSocket serverSocket = new DatagramSocket

(6428)

transport

application

physical

link

network

P3transport

application

physical

link

network

P1

transport

application

physical

link

network

P4

DatagramSocket mySocket1 = new DatagramSocket (5775)

DatagramSocket mySocket2 = new DatagramSocket (9157)

source port 9157dest port 6428

source port 6428dest port 9157

source port dest port

source port dest port

Transport Layer 3-12

Connection-oriented demux

TCP socket identified by 4-tuple source IP address source port number dest IP address dest port number

demux receiver uses all four values to direct segment to appropriate socket

server host may support many simultaneous TCP sockets each socket identified

by its own 4-tuple web servers have

different sockets for each connecting client non-persistent HTTP

will have different socket for each request

Transport Layer 3-13

Connection-oriented demux example

transport

application

physical

link

network

P3transport

application

physical

link

P4

transport

application

physical

link

network

P2

source IPport A9157dest IP port B80

source IPport B80dest IPport A9157

host IP address

A

host IP address

C

network

P6P5P3

source IPport C5775dest IPport B80

source IPport C9157dest IPport B80

three segments all destined to IP address B dest port 80 are demultiplexed to different sockets

server IP

address B

Transport Layer 3-14

Connection-oriented demux example

transport

application

physical

link

network

P3transport

application

physical

link

transport

application

physical

link

network

P2

source IPport A9157dest IP port B80

source IPport B80dest IPport A9157

host IP address

A

host IP address

C

server IP

address B

network

P3

source IPport C5775dest IPport B80

source IPport C9157dest IPport B80

P4

threaded server

Transport Layer 3-15

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-16

UDP User Datagram Protocol [RFC 768]

ldquono frillsrdquo ldquobare bonesrdquo Internet transport protocol

ldquobest effortrdquo service UDP segments may be lost delivered out-of-order

to app connectionless

no handshaking between UDP sender receiver

each UDP segment handled independently of others

UDP use streaming

multimedia apps (loss tolerant rate sensitive)

DNS SNMP

reliable transfer over UDP add reliability at

application layer application-specific

error recovery

Transport Layer 3-17

UDP segment header

source port dest port

32 bits

applicationdata

(payload)

UDP segment format

length checksum

length in bytes of UDP segment

including header

no connection establishment (which can add delay)

simple no connection state at sender receiver

small header size no congestion control

UDP can blast away as fast as desired

why is there a UDP

Transport Layer 3-18

UDP checksum

sender treat segment contents

including header fields as sequence of 16-bit integers

checksum addition (onersquos complement sum) of segment contents

sender puts checksum value into UDP checksum field

receiver compute checksum of

received segment check if computed

checksum equals checksum field value NO - error detected YES - no error detected

But maybe errors nonetheless More later hellip

Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment

Transport Layer 3-19

Internet checksum exampleexample add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Note when adding numbers a carryout from the most significant bit needs to be added to the

result

Transport Layer 3-20

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-21

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

Transport Layer 3-23

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

important in application transport link layers top-10 list of important networking topics

Principles of reliable data transfer

Transport Layer 3-24

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-25

wersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt)

consider only unidirectional data transfer but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by

next event

eventactions

Reliable data transfer getting started

Transport Layer 3-26

rdt10 reliable transfer over a reliable channel underlying channel perfectly reliable

no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver reads data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-27

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

rdt20 channel with bit errors

How do humans recover from ldquoerrorsrdquoduring conversation

Transport Layer 3-28

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection feedback control msgs (ACKNAK) from

receiver to sender

rdt20 channel with bit errors

Transport Layer 3-29

rdt20 FSM specification

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

belowsender

receiverrdt_send(data)

Transport Layer 3-30

rdt20 operation with no errors

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-31

rdt20 error scenario

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-32

rdt20 has a fatal flaw

what happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

canrsquot just retransmit possible duplicate

handling duplicates sender retransmits current

pkt if ACKNAK corrupted sender adds sequence

number to each pkt receiver discards (doesnrsquot

deliver up) duplicate pkt

stop and waitsender sends one packet then waits for receiver response

Transport Layer 3-33

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-34

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt21 receiver handles garbled ACKNAKs

Transport Layer 3-35

rdt21 discussion

sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquoexpectedrdquo pkt should have seq of 0 or 1

receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-36

rdt22 a NAK-free protocol

same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last

pkt received OK receiver must explicitly include seq of pkt being

ACKed duplicate ACK at sender results in same

action as NAK retransmit current pkt

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-11

Connectionless demux example

DatagramSocket serverSocket = new DatagramSocket

(6428)

transport

application

physical

link

network

P3transport

application

physical

link

network

P1

transport

application

physical

link

network

P4

DatagramSocket mySocket1 = new DatagramSocket (5775)

DatagramSocket mySocket2 = new DatagramSocket (9157)

source port 9157dest port 6428

source port 6428dest port 9157

source port dest port

source port dest port

Transport Layer 3-12

Connection-oriented demux

TCP socket identified by 4-tuple source IP address source port number dest IP address dest port number

demux receiver uses all four values to direct segment to appropriate socket

server host may support many simultaneous TCP sockets each socket identified

by its own 4-tuple web servers have

different sockets for each connecting client non-persistent HTTP

will have different socket for each request

Transport Layer 3-13

Connection-oriented demux example

transport

application

physical

link

network

P3transport

application

physical

link

P4

transport

application

physical

link

network

P2

source IPport A9157dest IP port B80

source IPport B80dest IPport A9157

host IP address

A

host IP address

C

network

P6P5P3

source IPport C5775dest IPport B80

source IPport C9157dest IPport B80

three segments all destined to IP address B dest port 80 are demultiplexed to different sockets

server IP

address B

Transport Layer 3-14

Connection-oriented demux example

transport

application

physical

link

network

P3transport

application

physical

link

transport

application

physical

link

network

P2

source IPport A9157dest IP port B80

source IPport B80dest IPport A9157

host IP address

A

host IP address

C

server IP

address B

network

P3

source IPport C5775dest IPport B80

source IPport C9157dest IPport B80

P4

threaded server

Transport Layer 3-15

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-16

UDP User Datagram Protocol [RFC 768]

ldquono frillsrdquo ldquobare bonesrdquo Internet transport protocol

ldquobest effortrdquo service UDP segments may be lost delivered out-of-order

to app connectionless

no handshaking between UDP sender receiver

each UDP segment handled independently of others

UDP use streaming

multimedia apps (loss tolerant rate sensitive)

DNS SNMP

reliable transfer over UDP add reliability at

application layer application-specific

error recovery

Transport Layer 3-17

UDP segment header

source port dest port

32 bits

applicationdata

(payload)

UDP segment format

length checksum

length in bytes of UDP segment

including header

no connection establishment (which can add delay)

simple no connection state at sender receiver

small header size no congestion control

UDP can blast away as fast as desired

why is there a UDP

Transport Layer 3-18

UDP checksum

sender treat segment contents

including header fields as sequence of 16-bit integers

checksum addition (onersquos complement sum) of segment contents

sender puts checksum value into UDP checksum field

receiver compute checksum of

received segment check if computed

checksum equals checksum field value NO - error detected YES - no error detected

But maybe errors nonetheless More later hellip

Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment

Transport Layer 3-19

Internet checksum exampleexample add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Note when adding numbers a carryout from the most significant bit needs to be added to the

result

Transport Layer 3-20

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-21

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

Transport Layer 3-23

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

important in application transport link layers top-10 list of important networking topics

Principles of reliable data transfer

Transport Layer 3-24

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-25

wersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt)

consider only unidirectional data transfer but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by

next event

eventactions

Reliable data transfer getting started

Transport Layer 3-26

rdt10 reliable transfer over a reliable channel underlying channel perfectly reliable

no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver reads data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-27

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

rdt20 channel with bit errors

How do humans recover from ldquoerrorsrdquoduring conversation

Transport Layer 3-28

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection feedback control msgs (ACKNAK) from

receiver to sender

rdt20 channel with bit errors

Transport Layer 3-29

rdt20 FSM specification

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

belowsender

receiverrdt_send(data)

Transport Layer 3-30

rdt20 operation with no errors

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-31

rdt20 error scenario

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-32

rdt20 has a fatal flaw

what happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

canrsquot just retransmit possible duplicate

handling duplicates sender retransmits current

pkt if ACKNAK corrupted sender adds sequence

number to each pkt receiver discards (doesnrsquot

deliver up) duplicate pkt

stop and waitsender sends one packet then waits for receiver response

Transport Layer 3-33

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-34

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt21 receiver handles garbled ACKNAKs

Transport Layer 3-35

rdt21 discussion

sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquoexpectedrdquo pkt should have seq of 0 or 1

receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-36

rdt22 a NAK-free protocol

same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last

pkt received OK receiver must explicitly include seq of pkt being

ACKed duplicate ACK at sender results in same

action as NAK retransmit current pkt

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-12

Connection-oriented demux

TCP socket identified by 4-tuple source IP address source port number dest IP address dest port number

demux receiver uses all four values to direct segment to appropriate socket

server host may support many simultaneous TCP sockets each socket identified

by its own 4-tuple web servers have

different sockets for each connecting client non-persistent HTTP

will have different socket for each request

Transport Layer 3-13

Connection-oriented demux example

transport

application

physical

link

network

P3transport

application

physical

link

P4

transport

application

physical

link

network

P2

source IPport A9157dest IP port B80

source IPport B80dest IPport A9157

host IP address

A

host IP address

C

network

P6P5P3

source IPport C5775dest IPport B80

source IPport C9157dest IPport B80

three segments all destined to IP address B dest port 80 are demultiplexed to different sockets

server IP

address B

Transport Layer 3-14

Connection-oriented demux example

transport

application

physical

link

network

P3transport

application

physical

link

transport

application

physical

link

network

P2

source IPport A9157dest IP port B80

source IPport B80dest IPport A9157

host IP address

A

host IP address

C

server IP

address B

network

P3

source IPport C5775dest IPport B80

source IPport C9157dest IPport B80

P4

threaded server

Transport Layer 3-15

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-16

UDP User Datagram Protocol [RFC 768]

ldquono frillsrdquo ldquobare bonesrdquo Internet transport protocol

ldquobest effortrdquo service UDP segments may be lost delivered out-of-order

to app connectionless

no handshaking between UDP sender receiver

each UDP segment handled independently of others

UDP use streaming

multimedia apps (loss tolerant rate sensitive)

DNS SNMP

reliable transfer over UDP add reliability at

application layer application-specific

error recovery

Transport Layer 3-17

UDP segment header

source port dest port

32 bits

applicationdata

(payload)

UDP segment format

length checksum

length in bytes of UDP segment

including header

no connection establishment (which can add delay)

simple no connection state at sender receiver

small header size no congestion control

UDP can blast away as fast as desired

why is there a UDP

Transport Layer 3-18

UDP checksum

sender treat segment contents

including header fields as sequence of 16-bit integers

checksum addition (onersquos complement sum) of segment contents

sender puts checksum value into UDP checksum field

receiver compute checksum of

received segment check if computed

checksum equals checksum field value NO - error detected YES - no error detected

But maybe errors nonetheless More later hellip

Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment

Transport Layer 3-19

Internet checksum exampleexample add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Note when adding numbers a carryout from the most significant bit needs to be added to the

result

Transport Layer 3-20

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-21

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

Transport Layer 3-23

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

important in application transport link layers top-10 list of important networking topics

Principles of reliable data transfer

Transport Layer 3-24

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-25

wersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt)

consider only unidirectional data transfer but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by

next event

eventactions

Reliable data transfer getting started

Transport Layer 3-26

rdt10 reliable transfer over a reliable channel underlying channel perfectly reliable

no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver reads data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-27

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

rdt20 channel with bit errors

How do humans recover from ldquoerrorsrdquoduring conversation

Transport Layer 3-28

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection feedback control msgs (ACKNAK) from

receiver to sender

rdt20 channel with bit errors

Transport Layer 3-29

rdt20 FSM specification

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

belowsender

receiverrdt_send(data)

Transport Layer 3-30

rdt20 operation with no errors

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-31

rdt20 error scenario

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-32

rdt20 has a fatal flaw

what happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

canrsquot just retransmit possible duplicate

handling duplicates sender retransmits current

pkt if ACKNAK corrupted sender adds sequence

number to each pkt receiver discards (doesnrsquot

deliver up) duplicate pkt

stop and waitsender sends one packet then waits for receiver response

Transport Layer 3-33

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-34

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt21 receiver handles garbled ACKNAKs

Transport Layer 3-35

rdt21 discussion

sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquoexpectedrdquo pkt should have seq of 0 or 1

receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-36

rdt22 a NAK-free protocol

same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last

pkt received OK receiver must explicitly include seq of pkt being

ACKed duplicate ACK at sender results in same

action as NAK retransmit current pkt

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-13

Connection-oriented demux example

transport

application

physical

link

network

P3transport

application

physical

link

P4

transport

application

physical

link

network

P2

source IPport A9157dest IP port B80

source IPport B80dest IPport A9157

host IP address

A

host IP address

C

network

P6P5P3

source IPport C5775dest IPport B80

source IPport C9157dest IPport B80

three segments all destined to IP address B dest port 80 are demultiplexed to different sockets

server IP

address B

Transport Layer 3-14

Connection-oriented demux example

transport

application

physical

link

network

P3transport

application

physical

link

transport

application

physical

link

network

P2

source IPport A9157dest IP port B80

source IPport B80dest IPport A9157

host IP address

A

host IP address

C

server IP

address B

network

P3

source IPport C5775dest IPport B80

source IPport C9157dest IPport B80

P4

threaded server

Transport Layer 3-15

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-16

UDP User Datagram Protocol [RFC 768]

ldquono frillsrdquo ldquobare bonesrdquo Internet transport protocol

ldquobest effortrdquo service UDP segments may be lost delivered out-of-order

to app connectionless

no handshaking between UDP sender receiver

each UDP segment handled independently of others

UDP use streaming

multimedia apps (loss tolerant rate sensitive)

DNS SNMP

reliable transfer over UDP add reliability at

application layer application-specific

error recovery

Transport Layer 3-17

UDP segment header

source port dest port

32 bits

applicationdata

(payload)

UDP segment format

length checksum

length in bytes of UDP segment

including header

no connection establishment (which can add delay)

simple no connection state at sender receiver

small header size no congestion control

UDP can blast away as fast as desired

why is there a UDP

Transport Layer 3-18

UDP checksum

sender treat segment contents

including header fields as sequence of 16-bit integers

checksum addition (onersquos complement sum) of segment contents

sender puts checksum value into UDP checksum field

receiver compute checksum of

received segment check if computed

checksum equals checksum field value NO - error detected YES - no error detected

But maybe errors nonetheless More later hellip

Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment

Transport Layer 3-19

Internet checksum exampleexample add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Note when adding numbers a carryout from the most significant bit needs to be added to the

result

Transport Layer 3-20

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-21

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

Transport Layer 3-23

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

important in application transport link layers top-10 list of important networking topics

Principles of reliable data transfer

Transport Layer 3-24

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-25

wersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt)

consider only unidirectional data transfer but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by

next event

eventactions

Reliable data transfer getting started

Transport Layer 3-26

rdt10 reliable transfer over a reliable channel underlying channel perfectly reliable

no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver reads data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-27

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

rdt20 channel with bit errors

How do humans recover from ldquoerrorsrdquoduring conversation

Transport Layer 3-28

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection feedback control msgs (ACKNAK) from

receiver to sender

rdt20 channel with bit errors

Transport Layer 3-29

rdt20 FSM specification

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

belowsender

receiverrdt_send(data)

Transport Layer 3-30

rdt20 operation with no errors

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-31

rdt20 error scenario

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-32

rdt20 has a fatal flaw

what happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

canrsquot just retransmit possible duplicate

handling duplicates sender retransmits current

pkt if ACKNAK corrupted sender adds sequence

number to each pkt receiver discards (doesnrsquot

deliver up) duplicate pkt

stop and waitsender sends one packet then waits for receiver response

Transport Layer 3-33

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-34

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt21 receiver handles garbled ACKNAKs

Transport Layer 3-35

rdt21 discussion

sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquoexpectedrdquo pkt should have seq of 0 or 1

receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-36

rdt22 a NAK-free protocol

same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last

pkt received OK receiver must explicitly include seq of pkt being

ACKed duplicate ACK at sender results in same

action as NAK retransmit current pkt

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-14

Connection-oriented demux example

transport

application

physical

link

network

P3transport

application

physical

link

transport

application

physical

link

network

P2

source IPport A9157dest IP port B80

source IPport B80dest IPport A9157

host IP address

A

host IP address

C

server IP

address B

network

P3

source IPport C5775dest IPport B80

source IPport C9157dest IPport B80

P4

threaded server

Transport Layer 3-15

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-16

UDP User Datagram Protocol [RFC 768]

ldquono frillsrdquo ldquobare bonesrdquo Internet transport protocol

ldquobest effortrdquo service UDP segments may be lost delivered out-of-order

to app connectionless

no handshaking between UDP sender receiver

each UDP segment handled independently of others

UDP use streaming

multimedia apps (loss tolerant rate sensitive)

DNS SNMP

reliable transfer over UDP add reliability at

application layer application-specific

error recovery

Transport Layer 3-17

UDP segment header

source port dest port

32 bits

applicationdata

(payload)

UDP segment format

length checksum

length in bytes of UDP segment

including header

no connection establishment (which can add delay)

simple no connection state at sender receiver

small header size no congestion control

UDP can blast away as fast as desired

why is there a UDP

Transport Layer 3-18

UDP checksum

sender treat segment contents

including header fields as sequence of 16-bit integers

checksum addition (onersquos complement sum) of segment contents

sender puts checksum value into UDP checksum field

receiver compute checksum of

received segment check if computed

checksum equals checksum field value NO - error detected YES - no error detected

But maybe errors nonetheless More later hellip

Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment

Transport Layer 3-19

Internet checksum exampleexample add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Note when adding numbers a carryout from the most significant bit needs to be added to the

result

Transport Layer 3-20

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-21

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

Transport Layer 3-23

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

important in application transport link layers top-10 list of important networking topics

Principles of reliable data transfer

Transport Layer 3-24

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-25

wersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt)

consider only unidirectional data transfer but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by

next event

eventactions

Reliable data transfer getting started

Transport Layer 3-26

rdt10 reliable transfer over a reliable channel underlying channel perfectly reliable

no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver reads data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-27

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

rdt20 channel with bit errors

How do humans recover from ldquoerrorsrdquoduring conversation

Transport Layer 3-28

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection feedback control msgs (ACKNAK) from

receiver to sender

rdt20 channel with bit errors

Transport Layer 3-29

rdt20 FSM specification

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

belowsender

receiverrdt_send(data)

Transport Layer 3-30

rdt20 operation with no errors

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-31

rdt20 error scenario

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-32

rdt20 has a fatal flaw

what happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

canrsquot just retransmit possible duplicate

handling duplicates sender retransmits current

pkt if ACKNAK corrupted sender adds sequence

number to each pkt receiver discards (doesnrsquot

deliver up) duplicate pkt

stop and waitsender sends one packet then waits for receiver response

Transport Layer 3-33

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-34

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt21 receiver handles garbled ACKNAKs

Transport Layer 3-35

rdt21 discussion

sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquoexpectedrdquo pkt should have seq of 0 or 1

receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-36

rdt22 a NAK-free protocol

same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last

pkt received OK receiver must explicitly include seq of pkt being

ACKed duplicate ACK at sender results in same

action as NAK retransmit current pkt

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-15

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-16

UDP User Datagram Protocol [RFC 768]

ldquono frillsrdquo ldquobare bonesrdquo Internet transport protocol

ldquobest effortrdquo service UDP segments may be lost delivered out-of-order

to app connectionless

no handshaking between UDP sender receiver

each UDP segment handled independently of others

UDP use streaming

multimedia apps (loss tolerant rate sensitive)

DNS SNMP

reliable transfer over UDP add reliability at

application layer application-specific

error recovery

Transport Layer 3-17

UDP segment header

source port dest port

32 bits

applicationdata

(payload)

UDP segment format

length checksum

length in bytes of UDP segment

including header

no connection establishment (which can add delay)

simple no connection state at sender receiver

small header size no congestion control

UDP can blast away as fast as desired

why is there a UDP

Transport Layer 3-18

UDP checksum

sender treat segment contents

including header fields as sequence of 16-bit integers

checksum addition (onersquos complement sum) of segment contents

sender puts checksum value into UDP checksum field

receiver compute checksum of

received segment check if computed

checksum equals checksum field value NO - error detected YES - no error detected

But maybe errors nonetheless More later hellip

Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment

Transport Layer 3-19

Internet checksum exampleexample add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Note when adding numbers a carryout from the most significant bit needs to be added to the

result

Transport Layer 3-20

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-21

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

Transport Layer 3-23

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

important in application transport link layers top-10 list of important networking topics

Principles of reliable data transfer

Transport Layer 3-24

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-25

wersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt)

consider only unidirectional data transfer but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by

next event

eventactions

Reliable data transfer getting started

Transport Layer 3-26

rdt10 reliable transfer over a reliable channel underlying channel perfectly reliable

no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver reads data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-27

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

rdt20 channel with bit errors

How do humans recover from ldquoerrorsrdquoduring conversation

Transport Layer 3-28

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection feedback control msgs (ACKNAK) from

receiver to sender

rdt20 channel with bit errors

Transport Layer 3-29

rdt20 FSM specification

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

belowsender

receiverrdt_send(data)

Transport Layer 3-30

rdt20 operation with no errors

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-31

rdt20 error scenario

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-32

rdt20 has a fatal flaw

what happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

canrsquot just retransmit possible duplicate

handling duplicates sender retransmits current

pkt if ACKNAK corrupted sender adds sequence

number to each pkt receiver discards (doesnrsquot

deliver up) duplicate pkt

stop and waitsender sends one packet then waits for receiver response

Transport Layer 3-33

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-34

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt21 receiver handles garbled ACKNAKs

Transport Layer 3-35

rdt21 discussion

sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquoexpectedrdquo pkt should have seq of 0 or 1

receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-36

rdt22 a NAK-free protocol

same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last

pkt received OK receiver must explicitly include seq of pkt being

ACKed duplicate ACK at sender results in same

action as NAK retransmit current pkt

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-16

UDP User Datagram Protocol [RFC 768]

ldquono frillsrdquo ldquobare bonesrdquo Internet transport protocol

ldquobest effortrdquo service UDP segments may be lost delivered out-of-order

to app connectionless

no handshaking between UDP sender receiver

each UDP segment handled independently of others

UDP use streaming

multimedia apps (loss tolerant rate sensitive)

DNS SNMP

reliable transfer over UDP add reliability at

application layer application-specific

error recovery

Transport Layer 3-17

UDP segment header

source port dest port

32 bits

applicationdata

(payload)

UDP segment format

length checksum

length in bytes of UDP segment

including header

no connection establishment (which can add delay)

simple no connection state at sender receiver

small header size no congestion control

UDP can blast away as fast as desired

why is there a UDP

Transport Layer 3-18

UDP checksum

sender treat segment contents

including header fields as sequence of 16-bit integers

checksum addition (onersquos complement sum) of segment contents

sender puts checksum value into UDP checksum field

receiver compute checksum of

received segment check if computed

checksum equals checksum field value NO - error detected YES - no error detected

But maybe errors nonetheless More later hellip

Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment

Transport Layer 3-19

Internet checksum exampleexample add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Note when adding numbers a carryout from the most significant bit needs to be added to the

result

Transport Layer 3-20

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-21

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

Transport Layer 3-23

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

important in application transport link layers top-10 list of important networking topics

Principles of reliable data transfer

Transport Layer 3-24

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-25

wersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt)

consider only unidirectional data transfer but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by

next event

eventactions

Reliable data transfer getting started

Transport Layer 3-26

rdt10 reliable transfer over a reliable channel underlying channel perfectly reliable

no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver reads data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-27

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

rdt20 channel with bit errors

How do humans recover from ldquoerrorsrdquoduring conversation

Transport Layer 3-28

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection feedback control msgs (ACKNAK) from

receiver to sender

rdt20 channel with bit errors

Transport Layer 3-29

rdt20 FSM specification

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

belowsender

receiverrdt_send(data)

Transport Layer 3-30

rdt20 operation with no errors

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-31

rdt20 error scenario

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-32

rdt20 has a fatal flaw

what happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

canrsquot just retransmit possible duplicate

handling duplicates sender retransmits current

pkt if ACKNAK corrupted sender adds sequence

number to each pkt receiver discards (doesnrsquot

deliver up) duplicate pkt

stop and waitsender sends one packet then waits for receiver response

Transport Layer 3-33

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-34

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt21 receiver handles garbled ACKNAKs

Transport Layer 3-35

rdt21 discussion

sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquoexpectedrdquo pkt should have seq of 0 or 1

receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-36

rdt22 a NAK-free protocol

same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last

pkt received OK receiver must explicitly include seq of pkt being

ACKed duplicate ACK at sender results in same

action as NAK retransmit current pkt

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-17

UDP segment header

source port dest port

32 bits

applicationdata

(payload)

UDP segment format

length checksum

length in bytes of UDP segment

including header

no connection establishment (which can add delay)

simple no connection state at sender receiver

small header size no congestion control

UDP can blast away as fast as desired

why is there a UDP

Transport Layer 3-18

UDP checksum

sender treat segment contents

including header fields as sequence of 16-bit integers

checksum addition (onersquos complement sum) of segment contents

sender puts checksum value into UDP checksum field

receiver compute checksum of

received segment check if computed

checksum equals checksum field value NO - error detected YES - no error detected

But maybe errors nonetheless More later hellip

Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment

Transport Layer 3-19

Internet checksum exampleexample add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Note when adding numbers a carryout from the most significant bit needs to be added to the

result

Transport Layer 3-20

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-21

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

Transport Layer 3-23

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

important in application transport link layers top-10 list of important networking topics

Principles of reliable data transfer

Transport Layer 3-24

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-25

wersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt)

consider only unidirectional data transfer but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by

next event

eventactions

Reliable data transfer getting started

Transport Layer 3-26

rdt10 reliable transfer over a reliable channel underlying channel perfectly reliable

no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver reads data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-27

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

rdt20 channel with bit errors

How do humans recover from ldquoerrorsrdquoduring conversation

Transport Layer 3-28

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection feedback control msgs (ACKNAK) from

receiver to sender

rdt20 channel with bit errors

Transport Layer 3-29

rdt20 FSM specification

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

belowsender

receiverrdt_send(data)

Transport Layer 3-30

rdt20 operation with no errors

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-31

rdt20 error scenario

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-32

rdt20 has a fatal flaw

what happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

canrsquot just retransmit possible duplicate

handling duplicates sender retransmits current

pkt if ACKNAK corrupted sender adds sequence

number to each pkt receiver discards (doesnrsquot

deliver up) duplicate pkt

stop and waitsender sends one packet then waits for receiver response

Transport Layer 3-33

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-34

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt21 receiver handles garbled ACKNAKs

Transport Layer 3-35

rdt21 discussion

sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquoexpectedrdquo pkt should have seq of 0 or 1

receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-36

rdt22 a NAK-free protocol

same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last

pkt received OK receiver must explicitly include seq of pkt being

ACKed duplicate ACK at sender results in same

action as NAK retransmit current pkt

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-18

UDP checksum

sender treat segment contents

including header fields as sequence of 16-bit integers

checksum addition (onersquos complement sum) of segment contents

sender puts checksum value into UDP checksum field

receiver compute checksum of

received segment check if computed

checksum equals checksum field value NO - error detected YES - no error detected

But maybe errors nonetheless More later hellip

Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment

Transport Layer 3-19

Internet checksum exampleexample add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Note when adding numbers a carryout from the most significant bit needs to be added to the

result

Transport Layer 3-20

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-21

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

Transport Layer 3-23

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

important in application transport link layers top-10 list of important networking topics

Principles of reliable data transfer

Transport Layer 3-24

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-25

wersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt)

consider only unidirectional data transfer but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by

next event

eventactions

Reliable data transfer getting started

Transport Layer 3-26

rdt10 reliable transfer over a reliable channel underlying channel perfectly reliable

no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver reads data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-27

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

rdt20 channel with bit errors

How do humans recover from ldquoerrorsrdquoduring conversation

Transport Layer 3-28

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection feedback control msgs (ACKNAK) from

receiver to sender

rdt20 channel with bit errors

Transport Layer 3-29

rdt20 FSM specification

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

belowsender

receiverrdt_send(data)

Transport Layer 3-30

rdt20 operation with no errors

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-31

rdt20 error scenario

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-32

rdt20 has a fatal flaw

what happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

canrsquot just retransmit possible duplicate

handling duplicates sender retransmits current

pkt if ACKNAK corrupted sender adds sequence

number to each pkt receiver discards (doesnrsquot

deliver up) duplicate pkt

stop and waitsender sends one packet then waits for receiver response

Transport Layer 3-33

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-34

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt21 receiver handles garbled ACKNAKs

Transport Layer 3-35

rdt21 discussion

sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquoexpectedrdquo pkt should have seq of 0 or 1

receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-36

rdt22 a NAK-free protocol

same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last

pkt received OK receiver must explicitly include seq of pkt being

ACKed duplicate ACK at sender results in same

action as NAK retransmit current pkt

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-19

Internet checksum exampleexample add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Note when adding numbers a carryout from the most significant bit needs to be added to the

result

Transport Layer 3-20

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-21

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

Transport Layer 3-23

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

important in application transport link layers top-10 list of important networking topics

Principles of reliable data transfer

Transport Layer 3-24

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-25

wersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt)

consider only unidirectional data transfer but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by

next event

eventactions

Reliable data transfer getting started

Transport Layer 3-26

rdt10 reliable transfer over a reliable channel underlying channel perfectly reliable

no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver reads data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-27

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

rdt20 channel with bit errors

How do humans recover from ldquoerrorsrdquoduring conversation

Transport Layer 3-28

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection feedback control msgs (ACKNAK) from

receiver to sender

rdt20 channel with bit errors

Transport Layer 3-29

rdt20 FSM specification

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

belowsender

receiverrdt_send(data)

Transport Layer 3-30

rdt20 operation with no errors

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-31

rdt20 error scenario

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-32

rdt20 has a fatal flaw

what happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

canrsquot just retransmit possible duplicate

handling duplicates sender retransmits current

pkt if ACKNAK corrupted sender adds sequence

number to each pkt receiver discards (doesnrsquot

deliver up) duplicate pkt

stop and waitsender sends one packet then waits for receiver response

Transport Layer 3-33

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-34

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt21 receiver handles garbled ACKNAKs

Transport Layer 3-35

rdt21 discussion

sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquoexpectedrdquo pkt should have seq of 0 or 1

receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-36

rdt22 a NAK-free protocol

same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last

pkt received OK receiver must explicitly include seq of pkt being

ACKed duplicate ACK at sender results in same

action as NAK retransmit current pkt

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-20

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-21

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

Transport Layer 3-23

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

important in application transport link layers top-10 list of important networking topics

Principles of reliable data transfer

Transport Layer 3-24

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-25

wersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt)

consider only unidirectional data transfer but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by

next event

eventactions

Reliable data transfer getting started

Transport Layer 3-26

rdt10 reliable transfer over a reliable channel underlying channel perfectly reliable

no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver reads data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-27

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

rdt20 channel with bit errors

How do humans recover from ldquoerrorsrdquoduring conversation

Transport Layer 3-28

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection feedback control msgs (ACKNAK) from

receiver to sender

rdt20 channel with bit errors

Transport Layer 3-29

rdt20 FSM specification

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

belowsender

receiverrdt_send(data)

Transport Layer 3-30

rdt20 operation with no errors

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-31

rdt20 error scenario

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-32

rdt20 has a fatal flaw

what happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

canrsquot just retransmit possible duplicate

handling duplicates sender retransmits current

pkt if ACKNAK corrupted sender adds sequence

number to each pkt receiver discards (doesnrsquot

deliver up) duplicate pkt

stop and waitsender sends one packet then waits for receiver response

Transport Layer 3-33

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-34

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt21 receiver handles garbled ACKNAKs

Transport Layer 3-35

rdt21 discussion

sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquoexpectedrdquo pkt should have seq of 0 or 1

receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-36

rdt22 a NAK-free protocol

same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last

pkt received OK receiver must explicitly include seq of pkt being

ACKed duplicate ACK at sender results in same

action as NAK retransmit current pkt

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-21

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

Transport Layer 3-23

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

important in application transport link layers top-10 list of important networking topics

Principles of reliable data transfer

Transport Layer 3-24

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-25

wersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt)

consider only unidirectional data transfer but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by

next event

eventactions

Reliable data transfer getting started

Transport Layer 3-26

rdt10 reliable transfer over a reliable channel underlying channel perfectly reliable

no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver reads data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-27

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

rdt20 channel with bit errors

How do humans recover from ldquoerrorsrdquoduring conversation

Transport Layer 3-28

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection feedback control msgs (ACKNAK) from

receiver to sender

rdt20 channel with bit errors

Transport Layer 3-29

rdt20 FSM specification

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

belowsender

receiverrdt_send(data)

Transport Layer 3-30

rdt20 operation with no errors

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-31

rdt20 error scenario

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-32

rdt20 has a fatal flaw

what happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

canrsquot just retransmit possible duplicate

handling duplicates sender retransmits current

pkt if ACKNAK corrupted sender adds sequence

number to each pkt receiver discards (doesnrsquot

deliver up) duplicate pkt

stop and waitsender sends one packet then waits for receiver response

Transport Layer 3-33

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-34

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt21 receiver handles garbled ACKNAKs

Transport Layer 3-35

rdt21 discussion

sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquoexpectedrdquo pkt should have seq of 0 or 1

receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-36

rdt22 a NAK-free protocol

same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last

pkt received OK receiver must explicitly include seq of pkt being

ACKed duplicate ACK at sender results in same

action as NAK retransmit current pkt

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-22

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Principles of reliable data transfer important in application transport link layers

top-10 list of important networking topics

Transport Layer 3-23

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

important in application transport link layers top-10 list of important networking topics

Principles of reliable data transfer

Transport Layer 3-24

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-25

wersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt)

consider only unidirectional data transfer but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by

next event

eventactions

Reliable data transfer getting started

Transport Layer 3-26

rdt10 reliable transfer over a reliable channel underlying channel perfectly reliable

no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver reads data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-27

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

rdt20 channel with bit errors

How do humans recover from ldquoerrorsrdquoduring conversation

Transport Layer 3-28

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection feedback control msgs (ACKNAK) from

receiver to sender

rdt20 channel with bit errors

Transport Layer 3-29

rdt20 FSM specification

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

belowsender

receiverrdt_send(data)

Transport Layer 3-30

rdt20 operation with no errors

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-31

rdt20 error scenario

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-32

rdt20 has a fatal flaw

what happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

canrsquot just retransmit possible duplicate

handling duplicates sender retransmits current

pkt if ACKNAK corrupted sender adds sequence

number to each pkt receiver discards (doesnrsquot

deliver up) duplicate pkt

stop and waitsender sends one packet then waits for receiver response

Transport Layer 3-33

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-34

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt21 receiver handles garbled ACKNAKs

Transport Layer 3-35

rdt21 discussion

sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquoexpectedrdquo pkt should have seq of 0 or 1

receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-36

rdt22 a NAK-free protocol

same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last

pkt received OK receiver must explicitly include seq of pkt being

ACKed duplicate ACK at sender results in same

action as NAK retransmit current pkt

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-23

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

important in application transport link layers top-10 list of important networking topics

Principles of reliable data transfer

Transport Layer 3-24

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-25

wersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt)

consider only unidirectional data transfer but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by

next event

eventactions

Reliable data transfer getting started

Transport Layer 3-26

rdt10 reliable transfer over a reliable channel underlying channel perfectly reliable

no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver reads data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-27

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

rdt20 channel with bit errors

How do humans recover from ldquoerrorsrdquoduring conversation

Transport Layer 3-28

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection feedback control msgs (ACKNAK) from

receiver to sender

rdt20 channel with bit errors

Transport Layer 3-29

rdt20 FSM specification

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

belowsender

receiverrdt_send(data)

Transport Layer 3-30

rdt20 operation with no errors

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-31

rdt20 error scenario

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-32

rdt20 has a fatal flaw

what happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

canrsquot just retransmit possible duplicate

handling duplicates sender retransmits current

pkt if ACKNAK corrupted sender adds sequence

number to each pkt receiver discards (doesnrsquot

deliver up) duplicate pkt

stop and waitsender sends one packet then waits for receiver response

Transport Layer 3-33

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-34

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt21 receiver handles garbled ACKNAKs

Transport Layer 3-35

rdt21 discussion

sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquoexpectedrdquo pkt should have seq of 0 or 1

receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-36

rdt22 a NAK-free protocol

same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last

pkt received OK receiver must explicitly include seq of pkt being

ACKed duplicate ACK at sender results in same

action as NAK retransmit current pkt

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-24

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-25

wersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt)

consider only unidirectional data transfer but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by

next event

eventactions

Reliable data transfer getting started

Transport Layer 3-26

rdt10 reliable transfer over a reliable channel underlying channel perfectly reliable

no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver reads data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-27

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

rdt20 channel with bit errors

How do humans recover from ldquoerrorsrdquoduring conversation

Transport Layer 3-28

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection feedback control msgs (ACKNAK) from

receiver to sender

rdt20 channel with bit errors

Transport Layer 3-29

rdt20 FSM specification

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

belowsender

receiverrdt_send(data)

Transport Layer 3-30

rdt20 operation with no errors

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-31

rdt20 error scenario

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-32

rdt20 has a fatal flaw

what happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

canrsquot just retransmit possible duplicate

handling duplicates sender retransmits current

pkt if ACKNAK corrupted sender adds sequence

number to each pkt receiver discards (doesnrsquot

deliver up) duplicate pkt

stop and waitsender sends one packet then waits for receiver response

Transport Layer 3-33

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-34

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt21 receiver handles garbled ACKNAKs

Transport Layer 3-35

rdt21 discussion

sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquoexpectedrdquo pkt should have seq of 0 or 1

receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-36

rdt22 a NAK-free protocol

same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last

pkt received OK receiver must explicitly include seq of pkt being

ACKed duplicate ACK at sender results in same

action as NAK retransmit current pkt

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-25

wersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt)

consider only unidirectional data transfer but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by

next event

eventactions

Reliable data transfer getting started

Transport Layer 3-26

rdt10 reliable transfer over a reliable channel underlying channel perfectly reliable

no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver reads data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-27

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

rdt20 channel with bit errors

How do humans recover from ldquoerrorsrdquoduring conversation

Transport Layer 3-28

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection feedback control msgs (ACKNAK) from

receiver to sender

rdt20 channel with bit errors

Transport Layer 3-29

rdt20 FSM specification

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

belowsender

receiverrdt_send(data)

Transport Layer 3-30

rdt20 operation with no errors

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-31

rdt20 error scenario

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-32

rdt20 has a fatal flaw

what happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

canrsquot just retransmit possible duplicate

handling duplicates sender retransmits current

pkt if ACKNAK corrupted sender adds sequence

number to each pkt receiver discards (doesnrsquot

deliver up) duplicate pkt

stop and waitsender sends one packet then waits for receiver response

Transport Layer 3-33

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-34

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt21 receiver handles garbled ACKNAKs

Transport Layer 3-35

rdt21 discussion

sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquoexpectedrdquo pkt should have seq of 0 or 1

receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-36

rdt22 a NAK-free protocol

same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last

pkt received OK receiver must explicitly include seq of pkt being

ACKed duplicate ACK at sender results in same

action as NAK retransmit current pkt

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-26

rdt10 reliable transfer over a reliable channel underlying channel perfectly reliable

no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver reads data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-27

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

rdt20 channel with bit errors

How do humans recover from ldquoerrorsrdquoduring conversation

Transport Layer 3-28

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection feedback control msgs (ACKNAK) from

receiver to sender

rdt20 channel with bit errors

Transport Layer 3-29

rdt20 FSM specification

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

belowsender

receiverrdt_send(data)

Transport Layer 3-30

rdt20 operation with no errors

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-31

rdt20 error scenario

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-32

rdt20 has a fatal flaw

what happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

canrsquot just retransmit possible duplicate

handling duplicates sender retransmits current

pkt if ACKNAK corrupted sender adds sequence

number to each pkt receiver discards (doesnrsquot

deliver up) duplicate pkt

stop and waitsender sends one packet then waits for receiver response

Transport Layer 3-33

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-34

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt21 receiver handles garbled ACKNAKs

Transport Layer 3-35

rdt21 discussion

sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquoexpectedrdquo pkt should have seq of 0 or 1

receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-36

rdt22 a NAK-free protocol

same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last

pkt received OK receiver must explicitly include seq of pkt being

ACKed duplicate ACK at sender results in same

action as NAK retransmit current pkt

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-27

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

rdt20 channel with bit errors

How do humans recover from ldquoerrorsrdquoduring conversation

Transport Layer 3-28

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection feedback control msgs (ACKNAK) from

receiver to sender

rdt20 channel with bit errors

Transport Layer 3-29

rdt20 FSM specification

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

belowsender

receiverrdt_send(data)

Transport Layer 3-30

rdt20 operation with no errors

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-31

rdt20 error scenario

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-32

rdt20 has a fatal flaw

what happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

canrsquot just retransmit possible duplicate

handling duplicates sender retransmits current

pkt if ACKNAK corrupted sender adds sequence

number to each pkt receiver discards (doesnrsquot

deliver up) duplicate pkt

stop and waitsender sends one packet then waits for receiver response

Transport Layer 3-33

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-34

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt21 receiver handles garbled ACKNAKs

Transport Layer 3-35

rdt21 discussion

sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquoexpectedrdquo pkt should have seq of 0 or 1

receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-36

rdt22 a NAK-free protocol

same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last

pkt received OK receiver must explicitly include seq of pkt being

ACKed duplicate ACK at sender results in same

action as NAK retransmit current pkt

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-28

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly

tells sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection feedback control msgs (ACKNAK) from

receiver to sender

rdt20 channel with bit errors

Transport Layer 3-29

rdt20 FSM specification

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

belowsender

receiverrdt_send(data)

Transport Layer 3-30

rdt20 operation with no errors

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-31

rdt20 error scenario

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-32

rdt20 has a fatal flaw

what happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

canrsquot just retransmit possible duplicate

handling duplicates sender retransmits current

pkt if ACKNAK corrupted sender adds sequence

number to each pkt receiver discards (doesnrsquot

deliver up) duplicate pkt

stop and waitsender sends one packet then waits for receiver response

Transport Layer 3-33

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-34

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt21 receiver handles garbled ACKNAKs

Transport Layer 3-35

rdt21 discussion

sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquoexpectedrdquo pkt should have seq of 0 or 1

receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-36

rdt22 a NAK-free protocol

same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last

pkt received OK receiver must explicitly include seq of pkt being

ACKed duplicate ACK at sender results in same

action as NAK retransmit current pkt

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-29

rdt20 FSM specification

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

belowsender

receiverrdt_send(data)

Transport Layer 3-30

rdt20 operation with no errors

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-31

rdt20 error scenario

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-32

rdt20 has a fatal flaw

what happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

canrsquot just retransmit possible duplicate

handling duplicates sender retransmits current

pkt if ACKNAK corrupted sender adds sequence

number to each pkt receiver discards (doesnrsquot

deliver up) duplicate pkt

stop and waitsender sends one packet then waits for receiver response

Transport Layer 3-33

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-34

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt21 receiver handles garbled ACKNAKs

Transport Layer 3-35

rdt21 discussion

sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquoexpectedrdquo pkt should have seq of 0 or 1

receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-36

rdt22 a NAK-free protocol

same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last

pkt received OK receiver must explicitly include seq of pkt being

ACKed duplicate ACK at sender results in same

action as NAK retransmit current pkt

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-30

rdt20 operation with no errors

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-31

rdt20 error scenario

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-32

rdt20 has a fatal flaw

what happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

canrsquot just retransmit possible duplicate

handling duplicates sender retransmits current

pkt if ACKNAK corrupted sender adds sequence

number to each pkt receiver discards (doesnrsquot

deliver up) duplicate pkt

stop and waitsender sends one packet then waits for receiver response

Transport Layer 3-33

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-34

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt21 receiver handles garbled ACKNAKs

Transport Layer 3-35

rdt21 discussion

sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquoexpectedrdquo pkt should have seq of 0 or 1

receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-36

rdt22 a NAK-free protocol

same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last

pkt received OK receiver must explicitly include seq of pkt being

ACKed duplicate ACK at sender results in same

action as NAK retransmit current pkt

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-31

rdt20 error scenario

Wait for call from above

snkpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Transport Layer 3-32

rdt20 has a fatal flaw

what happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

canrsquot just retransmit possible duplicate

handling duplicates sender retransmits current

pkt if ACKNAK corrupted sender adds sequence

number to each pkt receiver discards (doesnrsquot

deliver up) duplicate pkt

stop and waitsender sends one packet then waits for receiver response

Transport Layer 3-33

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-34

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt21 receiver handles garbled ACKNAKs

Transport Layer 3-35

rdt21 discussion

sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquoexpectedrdquo pkt should have seq of 0 or 1

receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-36

rdt22 a NAK-free protocol

same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last

pkt received OK receiver must explicitly include seq of pkt being

ACKed duplicate ACK at sender results in same

action as NAK retransmit current pkt

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-32

rdt20 has a fatal flaw

what happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

canrsquot just retransmit possible duplicate

handling duplicates sender retransmits current

pkt if ACKNAK corrupted sender adds sequence

number to each pkt receiver discards (doesnrsquot

deliver up) duplicate pkt

stop and waitsender sends one packet then waits for receiver response

Transport Layer 3-33

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-34

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt21 receiver handles garbled ACKNAKs

Transport Layer 3-35

rdt21 discussion

sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquoexpectedrdquo pkt should have seq of 0 or 1

receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-36

rdt22 a NAK-free protocol

same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last

pkt received OK receiver must explicitly include seq of pkt being

ACKed duplicate ACK at sender results in same

action as NAK retransmit current pkt

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-33

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-34

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt21 receiver handles garbled ACKNAKs

Transport Layer 3-35

rdt21 discussion

sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquoexpectedrdquo pkt should have seq of 0 or 1

receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-36

rdt22 a NAK-free protocol

same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last

pkt received OK receiver must explicitly include seq of pkt being

ACKed duplicate ACK at sender results in same

action as NAK retransmit current pkt

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-34

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt21 receiver handles garbled ACKNAKs

Transport Layer 3-35

rdt21 discussion

sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquoexpectedrdquo pkt should have seq of 0 or 1

receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-36

rdt22 a NAK-free protocol

same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last

pkt received OK receiver must explicitly include seq of pkt being

ACKed duplicate ACK at sender results in same

action as NAK retransmit current pkt

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-35

rdt21 discussion

sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquoexpectedrdquo pkt should have seq of 0 or 1

receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-36

rdt22 a NAK-free protocol

same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last

pkt received OK receiver must explicitly include seq of pkt being

ACKed duplicate ACK at sender results in same

action as NAK retransmit current pkt

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-36

rdt22 a NAK-free protocol

same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last

pkt received OK receiver must explicitly include seq of pkt being

ACKed duplicate ACK at sender results in same

action as NAK retransmit current pkt

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-37

rdt22 sender receiver fragments

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) || isACK(rcvpkt1) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

Wait for ACK

0

sender FSMfragment

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)

Wait for 0 from below

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) || has_seq1(rcvpkt))

udt_send(sndpkt)

receiver FSMfragment

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-38

rdt30 channels with errors and loss

new assumption underlying channel can also lose packets (data ACKs) checksum seq

ACKs retransmissions will be of help hellip but not enough

approach sender waits ldquoreasonablerdquo amount of time for ACK

retransmits if no ACK received in this time

if pkt (or ACK) just delayed (not lost) retransmission will be

duplicate but seq rsquos already handles this

receiver must specify seq of pkt being ACKed

requires countdown timer

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-39

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-40

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

pkt1

ack1

ack0

ack0

(a) no loss

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(b) packet loss

pkt1X

loss

pkt1timeout

resend pkt1

rdt30 in action

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-41

rdt30 in action

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

rcv pkt0

send ack0

send ack1

send ack0

rcv ack0

send pkt0

send pkt1

rcv ack1

send pkt0rcv pkt0

pkt0

pkt0

ack1

ack0

ack0

(c) ACK loss

ack1X

loss

pkt1timeout

resend pkt1

rcv pkt1send ack1

(detect duplicate)

pkt1

sender receiver

rcv pkt1

send ack0rcv ack0

send pkt1

send pkt0rcv pkt0

pkt0

ack0

(d) premature timeout delayed ACK

pkt1timeout

resend pkt1

ack1

send ack1

send pkt0rcv ack1

pkt0

ack1

ack0

send pkt0rcv ack1 pkt0

rcv pkt0send ack0ack0

rcv pkt0

send ack0(detect duplicate)

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-42

Performance of rdt30 rdt30 is correct but performance stinks eg 1 Gbps link 15 ms prop delay 8000 bit packet

U sender utilization ndash fraction of time sender busy sending

U sender =

008 30008

= 000027 L R

RTT + L R =

if RTT=30 msec 1KB pkt every 30 msec 33kBsec thruput over 1 Gbps link

network protocol limits use of physical resources

Dtrans =LR

8000 bits109 bitssec= = 8 microsecs

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-43

rdt30 stop-and-wait operation

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender =

008 30008

= 000027 L R

RTT + L R =

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-44

Pipelined protocols

pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be

increased buffering at sender andor receiver

two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-45

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

3-packet pipelining increases utilization by a factor of 3

U sender =

0024 30008

= 000081 3L R

RTT + L R =

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-46

Pipelined protocols overviewGo-back-N sender can have

up to N unacked packets in pipeline

receiver only sends cumulative ack doesnrsquot ack packet

if therersquos a gap sender has timer

for oldest unacked packet when timer expires

retransmit all unacked packets

Selective Repeat sender can have up

to N unackrsquoed packets in pipeline

rcvr sends individual ack for each packet

sender maintains timer for each unacked packet when timer expires

retransmit only that unacked packet

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-47

Go-Back-N sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts

allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for oldest in-flight pkt timeout(n) retransmit packet n and all higher seq

pkts in window

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-48

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-49

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) no receiver buffering re-ACK pkt with highest in-order seq

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

GBN receiver extended FSM

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-50

GBN in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 discard (re)send ack1rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5

Xloss

receive pkt4 discard (re)send ack1receive pkt5 discard (re)send ack1

rcv pkt2 deliver send ack2rcv pkt3 deliver send ack3rcv pkt4 deliver send ack4rcv pkt5 deliver send ack5

ignore duplicate ACK

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-51

Selective repeat receiver individually acknowledges all

correctly received pkts buffers pkts as needed for eventual in-

order delivery to upper layer sender only resends pkts for which

ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos limits seq s of sent unACKed pkts

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-52

Selective repeat sender receiver windows

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-53

Selective repeat

data from above if next available seq

in window send pkttimeout(n) resend pkt n restart

timerACK(n) in

[sendbasesendbase+N] mark pkt n as

received if n smallest unACKed

pkt advance window base to next unACKed seq

senderpkt n in [rcvbase

rcvbase+N-1] send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)otherwise ignore

receiver

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-54

Selective repeat in action

send pkt0send pkt1send pkt2send pkt3

(wait)

sender receiver

receive pkt0 send ack0receive pkt1 send ack1 receive pkt3 buffer send ack3rcv ack0 send pkt4

rcv ack1 send pkt5

pkt 2 timeoutsend pkt2

Xloss

receive pkt4 buffer send ack4receive pkt5 buffer send ack5

rcv pkt2 deliver pkt2pkt3 pkt4 pkt5 send ack2

record ack3 arrived

0 1 2 3 4 5 6 7 8

sender window (N=4)

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8

record ack4 arrived

record ack4 arrived

Q what happens when ack2 arrives

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-55

Selective repeatdilemma

example seq rsquos 0 1 2 3 window size=3

receiver window(after receipt)

sender window(after receipt)

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2 pkt0

timeoutretransmit pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2XXX

will accept packetwith seq number 0

(b) oops

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

pkt0

pkt1

pkt2

0 1 2 3 0 1 2pkt0

0 1 2 3 0 1 2

0 1 2 3 0 1 2

0 1 2 3 0 1 2

X

will accept packetwith seq number 0

0 1 2 3 0 1 2 pkt3

(a) no problem

receiver canrsquot see sender sidereceiver behavior identical in both cases

somethingrsquos (very) wrong

receiver sees no difference in two scenarios

duplicate data accepted as new in (b)

Q what relationship between seq size and window size to avoid problem in (b)

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-56

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-57

TCP Overview RFCs 79311221323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size connection-oriented

handshaking (exchange of control msgs) inits sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver reliable in-order

byte steam no ldquomessage

boundariesrdquo pipelined

TCP congestion and flow control set window size

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-58

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement number

receive window

Urg data pointerchecksum

FSRPAUheadlen

notused

options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-59

TCP seq numbers ACKssequence numbers

byte stream ldquonumberrdquo of first byte in segmentrsquos data

acknowledgementsseq of next byte expected from other side

cumulative ACKQ how receiver handles out-of-order segmentsA TCP spec doesnrsquot say - up to implementor

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

incoming segment to sender

A

sent ACKed

sent not-yet ACKed(ldquoin-flightrdquo)

usablebut not yet sent

not usable

window size N

sender sequence number space

source port dest port

sequence number

acknowledgement number

checksum

rwnd

urg pointer

outgoing segment from sender

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-60

TCP seq numbers ACKs

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo

simple telnet scenario

Host BHost A

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-61

TCP round trip time timeoutQ how to set TCP

timeout value longer than RTT

but RTT varies too short

premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to estimate RTT

SampleRTT measured time from segment transmission until ACK receipt ignore retransmissions

SampleRTT will vary want estimated RTT ldquosmootherrdquo average several

recent measurements not just current SampleRTT

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-62

RTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

EstimatedRTT = (1- )EstimatedRTT + SampleRTT exponential weighted moving average influence of past sample decreases

exponentially fast typical value = 0125

TCP round trip time timeout

RTT

(mill

iseco

nds)

RTT gaiacsumassedu to fantasiaeurecomfr

sampleRTT

EstimatedRTT

time (seconds)

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-63

timeout interval EstimatedRTT plus ldquosafety marginrdquo large variation in EstimatedRTT -gt larger safety margin

estimate SampleRTT deviation from EstimatedRTT

DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|

TCP round trip time timeout

(typically = 025)

TimeoutInterval = EstimatedRTT + 4DevRTT

estimated RTT ldquosafety marginrdquo

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-64

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-65

TCP reliable data transfer TCP creates rdt

service on top of IPrsquos unreliable service pipelined segments cumulative acks single

retransmission timer retransmissions

triggered by timeout events duplicate acks

letrsquos initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-66

TCP sender eventsdata rcvd from app create segment with

seq seq is byte-

stream number of first data byte in segment

start timer if not already running think of timer as for

oldest unacked segment

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer ack rcvd if ack acknowledges

previously unacked segments update what is

known to be ACKed start timer if there

are still unacked segments

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-67

TCP sender (simplified)

waitfor

event

NextSeqNum = InitialSeqNumSendBase = InitialSeqNum

create segment seq NextSeqNumpass segment to IP (ie ldquosendrdquo)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

data received from application above

retransmit not-yet-acked segment with smallest seq start timer

timeout

if (y gt SendBase) SendBase = y SendBasendash1 last cumulatively ACKed byte if (there are currently not-yet-acked segments) start timer else stop timer

ACK received with ACK field value y

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-68

TCP retransmission scenarios

lost ACK scenario

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8 bytes of data

Xtim

eo

ut

ACK=100

premature timeout

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=92 8bytes of data

tim

eo

ut

ACK=120

Seq=100 20 bytes of data

ACK=120

SendBase=100

SendBase=120

SendBase=120

SendBase=92

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-69

TCP retransmission scenarios

X

cumulative ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

Seq=120 15 bytes of data

tim

eo

ut

Seq=100 20 bytes of data

ACK=120

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-70

TCP ACK generation [RFC 1122 RFC

2581]

event at receiver

arrival of in-order segment withexpected seq All data up toexpected seq already ACKed

arrival of in-order segment withexpected seq One other segment has ACK pending

arrival of out-of-order segmenthigher-than-expect seq Gap detected

arrival of segment that partially or completely fills gap

TCP receiver action

delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK

immediately send single cumulative ACK ACKing both in-order segments

immediately send duplicate ACK indicating seq of next expected byte

immediate send ACK provided thatsegment starts at lower end of gap

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-71

TCP fast retransmit time-out period

often relatively long long delay before

resending lost packet detect lost

segments via duplicate ACKs sender often sends

many segments back-to-back

if segment is lost there will likely be many duplicate ACKs

if sender receives 3 ACKs for same data(ldquotriple duplicate ACKsrdquo) resend unacked segment with smallest seq

likely that unacked segment lost so donrsquot wait for timeout

TCP fast retransmit

(ldquotriple duplicate ACKsrdquo)

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-72

X

fast retransmit after sender receipt of triple duplicate ACK

Host BHost A

Seq=92 8 bytes of data

ACK=100

tim

eo

ut ACK=100

ACK=100

ACK=100

TCP fast retransmit

Seq=100 20 bytes of data

Seq=100 20 bytes of data

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-73

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-74

TCP flow controlapplication

process

TCP socketreceiver buffers

TCPcode

IPcode

application

OS

receiver protocol stack

application may remove data from

TCP socket buffers hellip

hellip slower than TCP

receiver is delivering(sender is sending)

from sender

receiver controls sender so sender wonrsquot overflow receiverrsquos buffer by transmitting too much too fast

flow control

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-75

TCP flow control

buffered data

free buffer spacerwnd

RcvBuffer

TCP segment payloads

to application process

receiver ldquoadvertisesrdquo free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via

socket options (typical default is 4096 bytes)

many operating systems autoadjust RcvBuffer

sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value

guarantees receive buffer will not overflow

receiver-side buffering

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-76

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-77

Connection Managementbefore exchanging data senderreceiver

ldquohandshakerdquo agree to establish connection (each knowing the

other willing to establish connection) agree on connection parameters

connection state ESTABconnection variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

connection state ESTABconnection Variables

seq client-to-server server-to-clientrcvBuffer size at serverclient

application

network

Socket clientSocket = newSocket(hostnameport

number)

Socket connectionSocket = welcomeSocketaccept()

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-78

Q will 2-way handshake always work in network

variable delays retransmitted messages

(eg req_conn(x)) due to message loss

message reordering canrsquot ldquoseerdquo other side

2-way handshake

Letrsquos talk

OKESTAB

ESTAB

choose xreq_conn(x)

ESTAB

ESTABacc_conn(x)

Agreeing to establish a connection

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-79

Agreeing to establish a connection

2-way handshake failure scenarios

retransmitreq_conn(

x)

ESTAB

req_conn(x)

half open connection(no client)

client terminat

es

serverforgets x

connection x completes

retransmitreq_conn(

x)

ESTAB

req_conn(x)

data(x+1)

retransmitdata(x+1)

acceptdata(x+1)

choose xreq_conn(x)

ESTAB

ESTAB

acc_conn(x)

client terminat

es

ESTAB

choose xreq_conn(x)

ESTAB

acc_conn(x)

data(x+1) acceptdata(x+1)

connection x completes server

forgets x

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-80

TCP 3-way handshake

SYNbit=1 Seq=x

choose init seq num xsend TCP SYN msg

ESTAB

SYNbit=1 Seq=yACKbit=1 ACKnum=x+1

choose init seq num ysend TCP SYNACKmsg acking SYN

ACKbit=1 ACKnum=y+1

received SYNACK(x) indicates server is livesend ACK for SYNACK

this segment may contain client-to-server data

received ACK(y) indicates client is live

SYNSENT

ESTAB

SYN RCVD

client state

LISTEN

server state

LISTEN

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-81

TCP 3-way handshake FSM

closed

listen

SYNrcvd

SYNsent

ESTAB

Socket clientSocket = newSocket(hostnameport

number)

SYN(seq=x)

Socket connectionSocket = welcomeSocketaccept()

SYN(x)

SYNACK(seq=yACKnum=x+1)create new socket for

communication back to client

SYNACK(seq=yACKnum=x+1)

ACK(ACKnum=y+1)ACK(ACKnum=y+1)

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-82

TCP closing a connection client server each close their side of

connection send TCP segment with FIN bit = 1

respond to received FIN with ACK on receiving FIN ACK can be combined with

own FIN simultaneous FIN exchanges can be

handled

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-83

FIN_WAIT_2

CLOSE_WAIT

FINbit=1 seq=y

ACKbit=1 ACKnum=y+1

ACKbit=1 ACKnum=x+1 wait for server

close

can stillsend data

can no longersend data

LAST_ACK

CLOSED

TIMED_WAIT

timed wait for 2max

segment lifetime

CLOSED

TCP closing a connection

FIN_WAIT_1 FINbit=1 seq=xcan no longersend but can receive data

clientSocketclose()

client state server state

ESTABESTAB

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-84

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-85

congestion informally ldquotoo many sources sending

too much data too fast for network to handlerdquo

different from flow control manifestations

lost packets (buffer overflow at routers)

long delays (queueing in router buffers)

a top-10 problem

Principles of congestion control

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-86

Causescosts of congestion scenario 1

two senders two receivers

one router infinite buffers

output link capacity R

no retransmission

maximum per-connection throughput R2

unlimited shared output link buffers

Host A

original data in

Host B

throughputout

R2

R2

out

in R2d

ela

yin

large delays as arrival rate in approaches capacity

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-87

one router finite buffers sender retransmission of timed-out packet

application-layer input = application-layer outputin

= out

transport-layer input includes retransmissions in in

finite shared output link buffers

Host A

in original data

Host B

outin original data plus

retransmitted data

lsquo

Causescosts of congestion scenario 2

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-88

idealization perfect knowledge

sender sends only when router buffers available

finite shared output link buffers

in original dataoutin original data plus

retransmitted data

copy

free buffer space

R2

R2

out

in

Causescosts of congestion scenario 2

Host B

A

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-89

in original dataoutin original data plus

retransmitted data

copy

no buffer space

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

Causescosts of congestion scenario 2

A

Host B

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-90

in original dataoutin original data plus

retransmitted data

free buffer space

Causescosts of congestion scenario 2

Idealization known loss packets can be lost dropped at router due to full buffers

sender only resends if packet known to be lost

R2

R2in

out

when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)

A

Host B

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-91

A

in outincopy

free buffer space

timeout

R2

R2in

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

Host B

Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Causescosts of congestion scenario 2

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-92

R2

out

when sending at R2 some packets are retransmissions including duplicated that are delivered

ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple

copies of pkt decreasing goodput

R2in

Causescosts of congestion scenario 2 Realistic duplicates packets can be lost

dropped at router due to full buffers

sender times out prematurely sending two copies both of which are delivered

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-93

four senders multihop paths timeoutretransmit

Q what happens as in and in

rsquo increase

finite shared output link buffers

Host A out

Causescosts of congestion scenario 3

Host B

Host CHost D

in original data

in original data plus

retransmitted data

A as red inrsquo increases all

arriving blue pkts at upper queue are dropped blue throughput 0

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-94

another ldquocostrdquo of congestion when packet dropped any ldquoupstream

transmission capacity used for that packet was wasted

Causescosts of congestion scenario 3

C2

C2

ou

t

inrsquo

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-95

Approaches towards congestion controltwo broad approaches towards congestion

controlend-end

congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

network-assisted congestion control

routers provide feedback to end systemssingle bit indicating congestion (SNA DECbit TCPIP ECN ATM)

explicit rate for sender to send at

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-96

Case study ATM ABR congestion control

ABR available bit rate

ldquoelastic servicerdquo if senderrsquos path

ldquounderloadedrdquo sender should

use available bandwidth

if senderrsquos path congested sender throttled

to minimum guaranteed rate

RM (resource management) cells

sent by sender interspersed with data cells

bits in RM cell set by switches (ldquonetwork-assistedrdquo) NI bit no increase in rate

(mild congestion) CI bit congestion

indication RM cells returned to sender

by receiver with bits intact

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-97

Case study ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sendersrsquo send rate thus max supportable rate on path

EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set receiver

sets CI bit in returned RM cell

RM cell data cell

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-98

Chapter 3 outline

31 transport-layer services

32 multiplexing and demultiplexing

33 connectionless transport UDP

34 principles of reliable data transfer

35 connection-oriented transport TCP segment structure reliable data transfer flow control connection

management36 principles of

congestion control37 TCP congestion

control

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-99

TCP congestion control additive increase multiplicative decrease

approach sender increases transmission rate (window size) probing for usable bandwidth until loss occurs additive increase increase cwnd by 1

MSS every RTT until loss detected multiplicative decrease cut cwnd in half

after loss

cwnd

TC

P s

ende

r co

nges

tion

win

dow

siz

e

AIMD saw toothbehavior probing

for bandwidth

additively increase window size helliphellip until loss occurs (then cut window in half)

time

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-100

TCP Congestion Control details

sender limits transmission

cwnd is dynamic function of perceived network congestion

TCP sending rate roughly send

cwnd bytes wait RTT for ACKS then send more bytes

last byteACKed sent not-

yet ACKed(ldquoin-flightrdquo)

last byte sent

cwnd

LastByteSent-LastByteAcked

lt cwnd

sender sequence number space

rate ~~cwnd

RTTbytessec

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-101

TCP Slow Start when connection

begins increase rate exponentially until first loss event initially cwnd = 1 MSS double cwnd every

RTT done by incrementing cwnd for every ACK received

summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RT

T

Host B

time

two segments

four segments

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-102

TCP detecting reacting to loss

loss indicated by timeout cwnd set to 1 MSS window then grows exponentially (as in slow start) to threshold

then grows linearly loss indicated by 3 duplicate ACKs TCP RENO

dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly

TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-103

Q when should the exponential increase switch to linear

A when cwnd gets to 12 of its value before timeout

Implementation variable ssthresh on loss event ssthresh is set to 12 of cwnd just before loss event

TCP switching from slow start to CA

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-104

Summary TCP Congestion Control

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd gt ssthresh

congestionavoidance

cwnd = cwnd + MSS (MSScwnd)dupACKcount = 0

transmit new segment(s) as allowed

new ACK

dupACKcount++

duplicate ACK

fastrecovery

cwnd = cwnd + MSStransmit new segment(s) as allowed

duplicate ACK

ssthresh= cwnd2cwnd = ssthresh + 3

retransmit missing segment

dupACKcount == 3

timeoutssthresh = cwnd2cwnd = 1 dupACKcount = 0retransmit missing segment

ssthresh= cwnd2cwnd = ssthresh + 3retransmit missing segment

dupACKcount == 3cwnd = ssthreshdupACKcount = 0

New ACK

slow start

timeoutssthresh = cwnd2

cwnd = 1 MSSdupACKcount = 0

retransmit missing segment

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s) as allowed

new ACKdupACKcount++

duplicate ACK

cwnd = 1 MSS

ssthresh = 64 KBdupACKcount = 0

NewACK

NewACK

NewACK

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-105

TCP throughput avg TCP thruput as function of window

size RTT ignore slow start assume always data to send

W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W avg thruput is 34W per RTT

W

W2

avg TCP thruput = 34W

RTTbytessec

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-106

TCP Futures TCP over ldquolong fat pipesrdquo example 1500 byte segments 100ms

RTT want 10 Gbps throughput requires W = 83333 in-flight segments throughput in terms of segment loss

probability L [Mathis 1997]

to achieve 10 Gbps throughput need a loss rate of L = 210-10 ndash a very small loss rate

new versions of TCP for high-speed

TCP throughput = 122 MSSRTT L

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-107

fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP Fairness

TCP connection 2

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-108

Why is TCP fairtwo competing sessions additive increase gives slope of 1 as throughout

increases multiplicative decrease decreases throughput

proportionally R

R

equal bandwidth share

Connection 1 throughput

Con

nect

ion

2 th

roug

h pu t

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-109

Fairness (more)Fairness and UDP multimedia apps

often do not use TCP do not want rate

throttled by congestion control

instead use UDP send audiovideo

at constant rate tolerate packet loss

Fairness parallel TCP connections

application can open multiple parallel connections between two hosts

web browsers do this eg link of rate R with 9

existing connections new app asks for 1 TCP gets

rate R10 new app asks for 11 TCPs gets

R2

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary

Transport Layer 3-110

Chapter 3 summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation implementation in the Internet UDP TCP

next leaving the

network ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

• Slide 1
• Chapter 3 Transport Layer
• Chapter 3 outline
• Transport services and protocols
• Transport vs network layer
• Internet transport-layer protocols
• Slide 7
• Multiplexingdemultiplexing
• How demultiplexing works
• Connectionless demultiplexing
• Connectionless demux example
• Connection-oriented demux
• Connection-oriented demux example
• Slide 14
• Slide 15
• UDP User Datagram Protocol [RFC 768]
• UDP segment header
• UDP checksum
• Internet checksum example
• Slide 20
• Principles of reliable data transfer
• Slide 22
• Slide 23
• Reliable data transfer getting started
• Slide 25
• rdt10 reliable transfer over a reliable channel
• rdt20 channel with bit errors
• Slide 28
• rdt20 FSM specification
• rdt20 operation with no errors
• rdt20 error scenario
• rdt20 has a fatal flaw
• rdt21 sender handles garbled ACKNAKs
• rdt21 receiver handles garbled ACKNAKs
• rdt21 discussion
• rdt22 a NAK-free protocol
• rdt22 sender receiver fragments
• rdt30 channels with errors and loss
• rdt30 sender
• rdt30 in action
• Slide 41
• Performance of rdt30
• rdt30 stop-and-wait operation
• Pipelined protocols
• Pipelining increased utilization
• Pipelined protocols overview
• Go-Back-N sender
• GBN sender extended FSM
• GBN receiver extended FSM
• GBN in action
• Selective repeat
• Selective repeat sender receiver windows
• Slide 53
• Selective repeat in action
• Selective repeat dilemma
• Slide 56
• TCP Overview RFCs 79311221323 2018 2581
• TCP segment structure
• TCP seq numbers ACKs
• Slide 60
• TCP round trip time timeout
• Slide 62
• Slide 63
• Slide 64
• TCP reliable data transfer
• TCP sender events
• TCP sender (simplified)
• TCP retransmission scenarios
• Slide 69
• TCP ACK generation [RFC 1122 RFC 2581]
• TCP fast retransmit
• Slide 72
• Slide 73
• TCP flow control
• Slide 75
• Slide 76
• Connection Management
• Agreeing to establish a connection
• Slide 79
• TCP 3-way handshake
• TCP 3-way handshake FSM
• TCP closing a connection
• Slide 83
• Slide 84
• Principles of congestion control
• Causescosts of congestion scenario 1
• Causescosts of congestion scenario 2
• Slide 88
• Slide 89
• Slide 90
• Slide 91
• Slide 92
• Causescosts of congestion scenario 3
• Slide 94
• Approaches towards congestion control
• Case study ATM ABR congestion control
• Slide 97
• Slide 98
• TCP congestion control additive increase multiplicative decrease
• TCP Congestion Control details
• TCP Slow Start
• TCP detecting reacting to loss
• TCP switching from slow start to CA
• Summary TCP Congestion Control
• TCP throughput
• TCP Futures TCP over ldquolong fat pipesrdquo
• TCP Fairness
• Why is TCP fair
• Fairness (more)
• Chapter 3 summary