Slide Set 13: TCP

In this set....TCP Connection TerminationTCP State Transition DiagramFlow ControlHow does TCP control its sliding window ?

Connection TerminationNote that after the server receives the FIN_WAIT_1, it may still have messages -- thus, connection not yet closed.

DATA/ACKActive CloseFIN_WAIT1FIN_WAIT2CLOSE_WAITLAST_ACKCLOSEDFIN, Seq Num = MACK= M+1FIN, Seq Num = NACK N+1

TCP State Transitions Note: Retransmissions and Data Packet /ACK exchanges are not represented in the state transition diagram. Final Wait time needed to ensure that the ACK is not lost. Simultaneous Connection Inceptions/ Terminations possible !

An Simpler View of the Client SideCLOSEDTIME_WAITFIN_WAIT2FIN_WAIT1ESTABLISHEDSYN_SENTSYN (Send)Rcv. SYN+ACK, Send ACKSend FINRcv. ACK, Send NothingRcv. FIN, Send ACK120 secs

Simpler Server ModelCLOSEDLAST ACKCLOSE_WAITESTABLISHEDSYN_RCVDLISTENPassive OPEN, Create Listen socketRcv. SYN, Send SYN+ACKRCV ACKRcv. FIN, Send ACKSend FINRcv. ACK, Send nothing

More about TerminationApplications on both sides have to independently close their half of the connection.If one side does it, this means that this side has no data to send but it is willing to receive.In the TIME_WAIT state, a client waits for 2 X MSL (typically). During this time the socket cannot be reused. If ACK is lost, a new FIN may be forthcoming and this second FIN may be delayed.Thus, if a new connection uses the same connection i.e., the same port numbers, this FIN would initiate termination of later connection !

Sequence Numbers and ACKsHow does one set Sequence numbers ? Implicitly a number in every byte in the stream.If we have 500000 bytes and each segment = MSS and = 1000 bytes, SN of 1st segment = 0, SN of second segment = 1000 and so on.Note that ACK number is the number that the receiving host puts in -- indicates the next byte that it is expecting.ACKs are cumulative -- ACK up to all the bytes that are received.

Flow ControlFlow control ensures that the sender does not send at a rate that causes the receiver buffer to overflow.Note that flow control is end-to-end.

Buffers at End HostsSending buffer Maintains data sent but not ACKedData written by application but not sent.Receive bufferData that arrives out of orderData that is in correct order but not yet read by application.

Sender Side ViewFor now, let us forget SN wrap around.Three pointers are maintained, LastByteAcked, LastByteSent, LastByteWritten.LastByteAcked LastByteSentLastByteSent LastByteWrittenSending applicationLastByteWrittenTCPLastByteSentLastByteAcked(a)

Receiver Side ViewThree pointers maintained again.LastByteRead < NextByteExpectedNextByteExpected LastByteRcvd + 1 Receiving applicationLastByteReadTCPLastByteRcvdNextByteExpected

How is Flow Control done?Receiver advertises a window size to the sender based on the buffer size allocated for the connection.Remember the Advertised Window field in the TCP header ?Sender cannot have more than Advertised Window bytes of unacknowledged data.Remember -- buffers are of finite size - i.e., there is a MaxRcvBuffer and MaxSendBuffer.

Setting the Advertised WindowOn the TCP receive side, clearly, LastByteRcvd -LastByteRead MaxRcvBufferThus, it advertises the space left in the buffer i.e., Advertised Window = MaxRcvBuffer - (LastByteRcvd -LastByteRead)As more data arrives i.e., more received bytes than read bytes, LastByteRcvd increases and hence, Advertised Window reduces.

Sender Side ResponseAt the sender side, the TCP sender should ensure that: LastByteSent - LastByteAcked Advertised Window.Thus, we define what is called the Effective Window which limits the amount of data that TCP can send : Effective Window = Advertised Window - (LastByteSent - LastByteAcked)Note here that ACKing does not imply that the process has read the data!In order to prevent the overflow of the Send Side buffer: LastByteWritten - LastByteAcked MaxSendBufferIf application tries to write more, TCP blocks.

PersistencyWhat does one do when Advertised Window = 0 ?The sender will persist by sending 1 segment.Note that this segment may not be accepted by the receiver initially.But at some point, it would trigger a response that may contain a new Advertised window.

Sequence Number WraparoundTCP Sequence Number is 32 bits long.Advertised Window is 16 bits. Since 232 >> 2 X 216, it is almost impossible for the same sequence number to exist twice -- wrap around unlikely.In addition, MSL = 120 seconds to make sure that there is no wrap-around. Time-stamps may also be used.

How long should the time-out be ?Remember, TCP has to ensure reliability. So bytes need to be resent if there is no timely acknowledgement.How long should the sender wait ?It should be adaptive -- fluctuation in load on the network. If too short, false time-outs If too long, then poor rate of sending.Depends on round trip time estimation

RTT EstimationSimple mechanism could be:Send packet, record time T1When ACK is returned, record time = T2.T2 -T1 = Estimated RTT.To avoid fluctuations, estimated RTT is a weighted average of previous time and current sampleEstimated RTT = (1-a) Estimated RTT + a SampleRTTIn the original specification a = 0.125The Time out is set to 2 * RTT.

A problem When there are retransmissions, it is unclear if the ACK is for the original transmission or for a retransmission. How do we overcome this ?

The Karn Patridge AlgorithmTake SampleRTT measurements only for segments that have been sent once !This eliminates the possibility that wrong RTT estimates are factored into the estimation.Another change -- Each time TCP retransmits, it sets the next timeout to 2 X Last timeout --> This is called the Exponential Back-off (primarily for avoiding congestion).

Jacobson Karels AlgorithmThe main problem with the Karn/Patridge scheme is that it does not take into account the variation between RTT samples.New method proposed -- the Jacobson Karels Algorithm.Estimated RTT = Estimated RTT + d X Difference Difference = Sample RTT - Estimated RTTDeviation = Deviation + d (|Difference| - deviation)Timeout = m Estimated RTT + f deviation.The values of m and f are computed based on experience -- Typically m = 1 and f = 4.

Silly Window SyndromeSuppose a MSS worth of data is collected and advertised window is MSS/2.What should the sender do ? -- transmit half full segments or wait to send a full MSS when window opens ?Early implementations were aggressive -- transmit MSS/2.Aggressively doing this, would consistently result in small segment sizes -- called the Silly Window Syndrome.

Issues ..We cannot eliminate the possibility of small segments being sent.However, we can introduce methods to coalesce small chunks. Delaying ACKs -- receiver does not send ACKs as soon as it receives segments. How long to delay ? Not very clear.Ultimate solution falls to the sender -- when should I transmit ?

Nagles AlgorithmIf sender waits too long --> bad for interactive connections.If it does not wait long enough -- silly window syndrome.How to solve ?Timer -- clock based If both available data and Window MSS, send full segment.Else, if there is unACKed data in flight, buffer new data until ACK returns.Else, send new data now.Note -- Socket interface allows some applications to turn off Nagles algorithm by setting the TCP-NODELAY option.

TCP ThroughputIf a connection sends W segments of MSS size (in bytes) in RTT seconds, then, the throughput is defined as : W *MSS / RTT bytes/second.If there is a link of capacity R, if there are K connections, what we want is for each TCP connection to have a throughput = R/K.

Throughput (cont)If a TCP session goes through n links and if link j has a rate Rj and is shared by Kj connections, ideally the throughput = Rj/Kj.Thus, a connections end-to-end rate is r = min (R1/K1, R2/K2, .. Rj/Kj... Rn/Kn).In reality not so simple, some connections may be unable to use their share -- so the share may be higher.

Where are we ?We have covered Chapter 5 -- Sections 5.1 and 5.2.Whatever I left out from Section 5.2 is for self-study.

Where are we headed ?We will look at Congestion Control with TCP next time.Chapter 6 -- Sections 6.3 and 6.4.