downlink tcp proxy solutions over hsdpa with multiple data flow

7/27/2019 Downlink TCP Proxy Solutions Over HSDPA With Multiple Data Flow

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Downlink TCP Proxy Solutions over

HSDPA with Multiple Data Flow

DANIELE GIRELLA

Masters Degree Project

Stockholm, Sweden 2007


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Downlink TCP Proxy Solutions over

HSDPA with Multiple Data Flow

DANIELE GIRELLA

Masters Degree Project

February 2007

Automatic Control Group

School of Electrical Engineering


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Abstract

In recent years, several proxy solutions have been proposed to improve perfor-

mance of TCP over wireless. The wide popularity of this protocol has pushed

for its adoption also in communication contexts, the wireless systems, where

the protocol was not intended to be applied. This is the case of High Speed

Downlink Packet Data Access (HSDPA), which is an enhancement of third

generation wireless systems in that it provides control mechanisms to increase

system performance. Despite shorter end-to-end delays, and more reliable suc-

cessful packet transmission, improving solutions for TCP over HSDPA are still

necessary. The goal of this Master thesis project is to explore the possibility

of design TCP proxy solutions to enhance users data rates over HSDPA. As a

relevant part of our activity, we have implemented a TCP proxy solution over

HSDPA through a ns2 simulator environment, extending the EURANE simu-

lator. EURANE has been developed within the SEACORN European project,

and it introduces three additional nodes to existing UMTS modules for ns2: the

Radio Network Controller, the Base Station and the User Equipment. The func-

tionality of these additional nodes allow for the support of the new features

introduced by HSDPA. The extension of the EURANE simulator includes all of

these HSDPA new features, a proxy solution, as well as some TCP enhancingprotocols (such as Eifel). The simulator allows for performance comparison of

existing TCP solution over wireless, and the proxy we have studied in this the-

sis. An analysis of the effects of multi-user data flows over TCP performance

have been also addressed.

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Introduction

HSDPA (High Speed Downlink Packet Access) represents a new high-speed

data transfer feature whose aim is to empower UMTS downlink data rates.

The need of increasing downlink data rates is due to the spreading of new 3G

mobile services -such as web browsing, streaming live video, network gaming-

that require high downlink sources and short latency.

The impressive increase in data rate is achieved by implementing a fast and

complex channel control mechanism based upon short physical layer frames,

Adaptive Modulation and Coding (AMC), fast Hybrid-Automatic Repeat re-

Quest (H-ARQ) and fast scheduling.

The HSDPA functionalities define three new channel types: High-Speed Down-

link Shared Channel (HS-DSCH), High-Speed Shared Control Channel (HS-SCCH)

and High-Speed Dedicated Physical Control Channel (HS-DPCCH). HS-DSCH is

multiplexed both in time and in code. In HSDPA each TTI lasts 2 ms compared

to 10 ms (or more) of UMTS. This reduction of TTI size permits to achieve a

shorter round trip delay between the User Equipment and the Node B, and

improve the link adaptation rate and efficiency of the AMC.

The distinctive characteristic of3rd Generation wireless networks is packet

data services. The information provided by these services are, in the major-

ity of the cases, accessible on the Internet which for the almost entirety works

with TCP traffic. Thus, there is a wide interest in extending TCP application in

mobile and wireless networks. The main problem of extending TCP over wire-

less networks is that it has been designed for wired networks where packet

losses are almost negligible and where delays are mainly caused by conges-

tion. Instead, in wireless networks the main source of packet losses is the link

level errors of the radio channel, which may seriously degrade the achievable

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vi INTRODUCTION

throughput of the TCP protocol.

It is well known that the main problem with TCP over networks having

both wired and wireless links is that packet losses are mistaken by the TCP

sender as being due to network congestion. The consequences are that TCP

drops its transmission window and often experiences time out, resulting in

degraded throughput.

The proposals to optimize TCP for wireless links can be divided into three

categories: link layer, end-to-end and split connections.

Link layer solutions (as Snoop Protocol) try to reduce the error rate of the link

through some kind of retransmission mechanism. As the data rate of the wire-less link increase, there will be more time for multiple link level retransmis-

sions before timeout occurs at the TCP layer, making link layer solutions more

viable.

End-to-end solutions (as Eifel Protocol) try to modify the TCP implementation

at the sender and/or receiver and/or intermediate routers, or optimizing the

parameters used by the TCP connection to achieve good performance.

Split connections (as Proxy Solutions) try to separate the TCP used in the wire-

less link from the one used in the wired one. The optimization procedure can

be done separately on the wired and wireless part.

In Chapter 1 will be introduced the High-Speed Downlink Packet Access

concept and its main new features, such as the new channel types, the Adaptive

Modulation and Coding, the Hybrid Automatic Repeat reQuest and the fast

scheduling. In the last section will be introduced the proposed evolution for

HSDPA.

In Chapter 2 will be reported a TCP overview regarding the architecture of

this protocol, its problems over 3G networks and a short a description of some

TCP versions.

In Chapter 3 will be introduced some TCP enhancing solutions, such as

Eifel and Snoop protocols, proxy and flow aggregation solutions, and so on.

In Chapter 4, using the network simulator ns-2 and a HSDPA implemen-

tation called EURANE, a comparative study of all the above solutions in an

HSDPA scenario will be provided.


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Acknowledgements

A thesis is the result of several years of study and hard work. Each of these

years is marked by bad and good days, each of these days is marked by bad

and good moments. On this way, one meets a lot of people that, in one way or

another, influence our life at that time. Many people have been a part of my

graduate education, as teachers, friends, and workmates. To all of them I want

to say thank you.

First of all, I want to express my gratitude to my supervisor, Carlo Fischione,

for the guidance, the support, and the many highlighting meetings he has pro-

vided me during this work. Thanks also for proposing me this master thesis

project.

I am very grateful to my swedish examiner, Karl H. Johansson, and to my ital-

ian examiner, Fortunato Santucci, for putting their faith in me and for giving

me the opportunity of doing my thesis in a world-class research group as the

Automatic Control Group of KTH.

Thanks to Pablo Soldati for his willingness and for giving me countless and

priceless advices during all my stay in Sweden. Thanks also to Alberto Sper-

anzon for helping me when I was in trouble with some control systems and to

Niels Moller for helping me with ns2.

Now is the moment to thank all those that have left a mark on my life dur-

ing my five years of studying at University of LAquila. My first thought goes

to Marco Fiorenzi, the best workmate I could have wished for. He has always

spurred me on to do my best, and to do it in that moment. Marco has been a

perfect workmate, a fantastic fellow traveller during the months we stayed in

Sweden but, first of all, he has been a real friend. Is thanks to him if I am here

now and if I have already finished my studies. Thanks to Gianluca Colantoni

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viii ACKNOWLEDGEMENTS

for its priceless friendship, for all the amusing and unique moments we have

past together and for the large heart he has always demonstrated to posses. A

special thought goes to Maria Ranieri, whose role in my life during all these

years is hard to explain by words. The simplest thing I can say is that she

was there, always, and she has always given me much more than I deserved.

Thanks also to Davide and Matteo Pacifico for their support, their willingness

and their unique capacity to solve every kind of problem I had. I would also

like to thank Massimo Paglia. Massimo has been a competent workmate, a

wise interlocutor and an excellent companion for enjoying.

Finally, a special thanks to those closest to me. Arianna, who shared myhappiness, and made me happy. Thanks for the love, patience, understanding,

and for putting your unreserved confidence in me. Thanks for being a so spe-

cial person.

My last (but not least!) thought goes to my family. I want to thank my father

Gabriele, my mother Uliana, and my sister Silvia for their understanding, end-

less patience and encouragement when it was most required. Is only thanks to

them if I have achieved this goal, and is only thanks to them if am what I am

today.

Thank you to all.


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Contents

1 High Speed Downlink Packet Access (HSDPA) 1

1.1 HSDPA Concept . . . . . . . . . . . . . . . . . . . . . . . . . . . 1

1.2 Channel Structure . . . . . . . . . . . . . . . . . . . . . . . . . . . 2

1.3 New Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6

1.3.1 Adaptive Modulation and Coding . . . . . . . . . . . . . 6

1.3.2 Fast Hybrid Automatic Repeat reQuest . . . . . . . . . . 9

1.3.3 Fast Scheduling . . . . . . . . . . . . . . . . . . . . . . . . 11

1.4 Comparative Study of HSDPA and WCDMA . . . . . . . . . . . 121.5 Evolution of HSDPA . . . . . . . . . . . . . . . . . . . . . . . . . 13

2 TCP Overview 21

2.1 TCP Architecture . . . . . . . . . . . . . . . . . . . . . . . . . . . 21

2.2 TCP Problems over 3G Networks . . . . . . . . . . . . . . . . . . 26

2.3 TCP Versions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28

2.4 Round Trip Time and mean number of retransmissions for TCP

over 3G . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33

3 TCP Enhancing Solutions 37

3.1 Proxy Solution . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 37

3.2 Flow Aggregation . . . . . . . . . . . . . . . . . . . . . . . . . . . 39

3.3 Eifel Protocol . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 43

3.4 Snoop Protocol . . . . . . . . . . . . . . . . . . . . . . . . . . . . 46

3.5 Further Enhancing Protocols . . . . . . . . . . . . . . . . . . . . 47

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x CONTENTS

4 Simulation 53

4.1 ns-2 Simulator and EURANE extension . . . . . . . . . . . . . . 53

4.2 Simulation Scenario . . . . . . . . . . . . . . . . . . . . . . . . . . 60

4.3 Simulation Results . . . . . . . . . . . . . . . . . . . . . . . . . . 63

5 Conclusions 73

References 75


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List of Figures

1.1 HS-PDSCH channel time and code multiplexing . . . . . . . . . 3

1.2 HS-SCCH frame structure . . . . . . . . . . . . . . . . . . . . . . 4

1.3 HS-DPCCH frame structure [1] . . . . . . . . . . . . . . . . . . . 5

1.4 HSDPA channel functionality . . . . . . . . . . . . . . . . . . . . 5

1.5 HSDPA physical layer . . . . . . . . . . . . . . . . . . . . . . . . 6

1.6 HSDPA UE categories . . . . . . . . . . . . . . . . . . . . . . . . 9

1.7 IR and CC state diagrams . . . . . . . . . . . . . . . . . . . . . . 10

1.8 An example of Chase Combining retransmission . . . . . . . . . 11

1.9 An example of Incremental Redundancy retransmission . . . . . 11

1.10 HSUPA peak throughput rates . . . . . . . . . . . . . . . . . . . 15

2.1 TCP slow start and congestion avoidance phase . . . . . . . . . 24

2.2 TCP fast retransmit and fast recovery phase . . . . . . . . . . . . 25

2.3 Mean value Ns as a function of BLER [26] . . . . . . . . . . . . . 35

2.4 Variance 2 as a function of BLER [26] . . . . . . . . . . . . . . . 35

3.1 Proxy solution architecture . . . . . . . . . . . . . . . . . . . . . . 38

3.2 RNF signalling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 39

3.3 TCP flow aggregation scheme [30] . . . . . . . . . . . . . . . . . 40

3.4 Sample logical aggregate for a give Mobile Host [30] . . . . . . . 41

3.5 Eifel procedure [35]. . . . . . . . . . . . . . . . . . . . . . . . . . . 45

3.6 Snoop procedure [39] . . . . . . . . . . . . . . . . . . . . . . . . . 46

4.1 UE side MAC architecture [50] . . . . . . . . . . . . . . . . . . . 55

4.2 UTRAN side overall MAC architecture [50] . . . . . . . . . . . . 55

4.3 Main characteristics of EURANEs schedulers [52] . . . . . . . . 57

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xii LIST OF FIGURES

4.4 Overview of physical layer model used in EURANE [52] . . . . 58

4.5 Simulation scenario . . . . . . . . . . . . . . . . . . . . . . . . . . 60

4.6 Available link bandwidth . . . . . . . . . . . . . . . . . . . . . . 61

4.7 Network architecture . . . . . . . . . . . . . . . . . . . . . . . . . 61

4.8 UEs throughput in the simple scenario . . . . . . . . . . . . . . 63

4.9 Servers congestion window in simple and RNFProxy scenarios 64

4.10 Trends obtained in simple scenario setting servers cwnd to 19 . 65

4.11 Throughput improvements by adding Eifel and Snoop protocols 66

4.12 UEs throughput in RNFProxy scenario . . . . . . . . . . . . . . 68

4.13 Throughput improvements by adding Eifel and Snoop protocolsto RNFProxy scenario . . . . . . . . . . . . . . . . . . . . . . . . . 69

4.14 UEs throughput in RNFProxy scenario with both Eifel and Snoop

protocols . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 70

4.15 Comparison between throughputs trend in simple scenario and

in RNFProxy scenario . . . . . . . . . . . . . . . . . . . . . . . . . 70

4.16 Comparison between throughputs trend in RNFProxy scenario

(with and without enhancing protocols) . . . . . . . . . . . . . . 71

4.17 Comparison between the throughput experienced interposing a

RNFProxy and that experienced with a SimpleProxy . . . . . . . 71


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List of Tables

1.1 2G to 3G throughput comparison . . . . . . . . . . . . . . . . . . . . 2

1.2 Comparison between DSCH and HS-DSCH basic properties . . 13

2.1 TCP versions comparison . . . . . . . . . . . . . . . . . . . . . . 33

4.1 Scenarios characteristics . . . . . . . . . . . . . . . . . . . . . . . 61

4.2 Simulation parameters . . . . . . . . . . . . . . . . . . . . . . . . 62

4.3 Summary of simulation results . . . . . . . . . . . . . . . . . . . 72

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List of Abbreviations

3G Third Generation3GPP 3rd Generation Partnership Project

ACB Aggregate Control Block

ACK Acknowledgment

AMC Adaptive Modulation and Coding

ARQ Automatic Repeat Request

ATCP Aggregate TCP

BCH Broadcast Channel

BDP Bandwidth Delay Product

BER Bit Error Rate

BLER Block Error Rate

C/I Carrier to Interference Ratio

CC Chase Combining

CDMA Code Division Multiple Access

CQI Channel Quality Indicator

CRC Cyclic Redundancy Check

CWND Congestion window

DCH Dedicated Channel

DPCH Dedicated Physical Channel

DSCH Downlink Shared Channel

DUPACK Duplicate Acknowledgment

E-DCH Enhanced Dedicated Channel

EDGE Enhanced Data rates for Global Evolution

EURANE Enhanced UMTS Radio Access Network Extension

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xvi LIST OF ABBREVIATIONS

FACH Forward Access Channel

FACK Forward Acknowledgment

FDD Frequency Division Duplex

FH Fixed Host

GGSN Gateway GPRS Support Node

GPRS General Packet Radio Service

GSM Global System for Mobile communication

H-ARQ Hybrid Automatic Repeat Request

HSDPA High-Speed Downlink Packet Access

HS-DPCCH High-Speed Dedicated Physical Control Channel

HS-DSCH High-speed Downlink Shared Channel

HS-PDSCH High-Speed Physical Downlink Shared Channel

HS-SCCH High-Speed Shared Control Channel

HSPA High-Speed Packet Access

HSUPA High Speed Uplink Packet Access

IR Incremental Redundancy

LTE Long Term Evolution

MAC Medium Access ControlMAC-b Medium Access Control for BCH

MAC-c Medium Access Control for PCH

MAC-d Medium Access Control for DCH

MAC-hs Medium Access Control high-speed

MAC-sh Medium Access Control for DSCH

MCS Modulation and Coding Scheme

MH Mobile Host

MIMO Multiple Input Multiple Output

MSR Mobile Support Router

MSS Maximum Segment Size

OFDM Orthogonal Frequency Division Multiplexing

OFDMA Orthogonal Frequency Division Multiplexing Access

PCH Paging Channel

PF Proportional Fair

QAM Quadrature amplitude modulation


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LIST OF ABBREVIATIONS xvii

QPSK Quadrature Phase Shift Keying

RACH Random Access Channel

RLC Radio Link Control

RNC Radio Network Controller

RNF Radio Network Feedback

RR Round Robin

RTO Retransmission Timeout

RTT Round Trip Time

RWND Receiver Window

SACK Selective Acknowledgment

SAW Stop And Wait

SDMA Space Division Multiple Access

SF Spreading Factor

SGSN Serving GPRS Support Node

SH Supervisory Host

SIR Signal to Interference Ratio

SSTHRESH Slow Start Threshold

SYN SynchronizeTCP Transmission Control Protocol

TDD Time Division Duplex

TLE Transmission Layer Efficiency

TTI Transmission Time Interval

UDP User Datagram Protocol

UE User Equipment

UMTS Universal Mobile Telecommunication System

UTRAN UMTS Terrestrial Radio Access Network

VOIP Voice over IP

WCDMA Wideband Code Division Multiple Access

WLAN Wireless Local Area Network


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Chapter 1

High Speed Downlink PacketAccess (HSDPA)

1.1 HSDPA Concept

HSDPA (High Speed Downlink Packet Access) represents a new high-speed

data transfer feature released by the 3rd Generation Partnership Project (3GPP)

with the aim of empower UMTS downlink data rates. The need of increasing

downlink data rates is due to the spreading of new 3G mobile services - such

as web browsing, streaming live video, network gaming - which require high

downlink sources whereas uplink is used only for control signalling. HSDPA

offers a way to increase downlink capacity within the existing spectrum by a

factor 2:3 compared to 3G Release 99. In Table 1.1 a comparison among 2G (the

basic GSM), 2.5G (GPRS and EDGE) and 3G (UMTS Rel. 99 and HSDPA Rel. 5)

downloading data rates is shown.

Another important enhancement introduced by HSDPA is a three-to-five-

fold sector throughput increase, which means more data users on a single fre-

quency (or carrier). The impressive increase of data rate is achieved by im-

plementing a fast and complex channel control mechanism based upon short

physical layer frames (cf. sec. 1.2), Adaptive Modulation and Coding (AMC)

(cf. sec. 1.3.1), fast Hybrid-Automatic Repeat reQuest (H-ARQ) (cf. sec. 1.3.2)

and fast scheduling (cf. sec. 1.3.3).

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2 CHAPTER 1. High Speed Downlink Packet Access (HSDPA)

GSM GPRS EDGE UMTS HSDPA

Typical max.

data rate

9.6 kbps 40 kbps 120 kbps 384 kbps 0.910 Mbps

Theoret. peak

data rate

14.4 kbps 171 kbps 473 kbps 2 Mbps 14.4 Mbps

Table 1.1: 2G to 3G throughput comparison

It is important to note that HSDPA is a pure access evolution without any

core networks impacts, except for minor changes due to the higher bandwidth

access. For instance, in the 3GPP Rel. 5 the maximum throughput set into the

signalling protocol has been increased from 2 Mbps to 16 Mbps in order to

support the theoretical maximum limit of HSDPA data rate (14.4 Mbps). It

follows that the deployment of HSDPA is very cost effective since the incre-

mental cost is mainly due to Node B and Radio Network Controller (RNC)

hardware/software upgrade while the operator cost to provide data services is

significantly reduced. In a typical dense urban environment, the operator cost

to deliver a megabyte of data traffic is about three cents with HSDPA while it

increases to about seven cents for UMTS. This is due to the high improvements

in spectral efficiency introduced by HSDPA.

1.2 Channel Structure

The HSDPA functionality defines three new channel types (see Fig. 1.4):

- High-Speed Downlink Shared Channel (HS-DSCH)

- High-Speed Shared Control Channel (HS-SCCH)

- High-Speed Dedicated Physical Control Channel (HS-DPCCH)

HS-DSCH is very similar to the DSCH transport channel defined in Rel. 99.

HS-DSCH has been introduced in Rel. 5 as the primary radio bearer and its

resources can be shared between all active HSDPA users in the cell. To obtain


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1.2. Channel Structure 3

higher data rates and greater spectral efficiency, the fast power control and

variable spreading factor of the DSCH are replaced in Rel. 5 by short packet

size, multicode operation, and techniques such as AMC and HARQ on the HS-

DSCH. Another difference from DSCH is that the scheduling with HS-DSCH is

done at the Node B rather than RNC. The HS-DSCH is mapped onto a pool of

physical channels (i.e. channelization codes) denominated HS-PDSCHs (High

Speed Physical Downlink Shared Channel) to be shared among all the HSDPA

users on a time multiplexed manner. HS-PDSCHs are multiplexed both in time

and in code. In Rel. 5, timeslots have the same length as in Rel. 99 (0.67 ms) but

differently from the latter where each Transmission Time Interval (TTI) consistsof15 slots (i.e. each TTI lasts 10 ms), in HSDPA each TTI consists of three slots

(i.e. 2 ms). This reduction of TTI size permits to achieve a shorter round trip

delay between the User Equipment (UE) and the Node B, and improves the link

adaptation rate and efficiency of the AMC. Within each 2 ms TTI, a constant

Spreading Factor (SF) of16 is used with a maximum of15 parallel channels for

HS-PDSCHs. These channels may all be assigned to one user during the TTI,

or may be split among several users (see Figure 1.1).

SpreadingCodes

user 1 user 2 user 3 user 4 user 5

HSDPA frame = 2 ms

Standard Rel. 99 frame = 10 ms

Figure 1.1: HS-PDSCH channel time and code multiplexing

In order to support the HS-DSCH operation, an HSDPA UE needs new con-

trol channels: the HS-SCCH in the downlink direction and the HS-DPCCH in


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the uplink direction.

HS-SCCH isa fixed rate(60 Kbps, SF=128) channel used for carrying down-

link signaling between the Node B and the UE before the beginning of each

scheduled TTI. This channel indicates the UE when there is data on the HS-

DSCH that is addressed to that specific UE, and gives the UE the fast chang-

ing parameters that are needed for HS-DSCH reception. This includes HARQ-

related information and the parameters of the HS-DSCH transport format se-

lected by the link adaptation mechanism (see Figure 1.2).

T = 0.67 ms slot2 x T = 1.33 msslot

code set , UE idmodulation scheme ,

other information

Figure 1.2: HS-SCCH frame structure

HS-DPCCH (SF=256) is an uplink low bandwidth channel used to carry

both ACK/NACK signaling indicating whether the corresponding downlink

transmission was successfully decoded and Channel Quality Indicator (CQI)

used to achieve link adaptation. To aid the power control operation of the HS-DPCCH, an associated Dedicated Physical Channel (DPCH) is run for every

user (see Figure 1.3).

Figure 1.5 describes the downlink and uplink channel structure of HSDPA.


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1.2. Channel Structure 5

Figure 1.3: HS-DPCCH frame structure [1]

CQI ( HS-DPCCH )

Downlink Tranfer Information ( HS-SCCH )

Data Transfer ( HS-DSCH )

ACK / NACK (HS-DPCCH)

Figure 1.4: HSDPA channel functionality


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Slot (0.67 ms)

CQI ACK

Uplink

Downlink

DL associatedDPCH(for each HSDPA user)

HS-SCCH

HS-PDSCH #1

HS-PDSCH #15

UL associated DPCH(for each HSDPA user)

HS-DPCCH

TTI (2 ms)

7.5 slots

CQI CQI

HS-PDSCH #2

Figure 1.5: HSDPA physical layer

1.3 New Features

As mentioned in previous section, HSDPA introduces three new features:

Adaptive Modulation and Coding (AMC).

Hybrid Automatic Repeat reQuest (HARQ).

Fast Scheduling.

1.3.1 Adaptive Modulation and Coding

Adaptive Modulation and Coding (AMC) represents a fundamental feature of HS-DPA. It consists of continuously optimizing the modulation scheme, the code

rate, the number of codes employed and the transmit power per code. This

otpimization is based on various sources [2]:

Channel Quality Indicator (CQI): the UE sends in the uplink a report de-

nominated CQI that provides implicit information about the instanta-

neous signal quality received by the user. The CQI specifies the modula-

tion, the number of codes and the transport block size the UE can support


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1.3. New Features 7

with a detection error no higher than 10% [3]. This error is referred to the

first transmission and to a reference HS-PDSCH power. The RNC com-

mands the UE to report the CQI every 2, 4, 8, 10, 20, 40, 80 or 160 ms [4]

or to disable the report. In [3], the complete set of reference CQI reports

is defined.

Power Measurements on the Associated DPCH: every user to be mapped

on to HS-PDSCH runs a parallel DPCH for signalling purposes, whose

transmission power can be used to gain knowledge about the instan-

taneous status of the users channel quality. This information may be

employed for link adaptation [5] as well as for packet scheduling. The

advantages of using this information are that no additional signalling is

required, and that it is available on a slot basis. However, it is limited to

the case when the HS-DSCH and the DPCH apply the same type of detec-

tor (e.g. a conventional Rake), and can not be used when the associated

DPCH enters soft handover.

Hybrid ARQ Acknowledgements: the acknowledgement corresponding to

the HARQ protocol may provide an estimation of the users channel

quality too, although this information is expected to be less frequent than

previous ones because it is only received when the user is served. Hence,

it does not provide instantaneous channel quality information. Note that

it also lacks the channel quality resolution provided by the two previous

metrics since a single information bit is reported.

Buffer Size: the amount of data in the Medium Access Control (MAC)

buffer could also be applied in combination with previous information

to select the transmission parameters.

HSDPA uses higher order modulation schemes as 16-quadrature amplitude

modulation (16-QAM) besides the existing QPSK used for Rel. 99 channels.

The modulation to be used is adapted according to the radio channel condi-

tions. The HS-DSCH encoding scheme is based on the Rel. 99 rate (1/3 turbo

encoder) but adds rate matching with puncturing and repetition to improve

the granularity of the effective code rate (1/4, 1/2, 5/8, 3/4). Different combi-

nations of modulation and channel coding-rate can be used to provide different


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peak data rates. In HSDPA, users close to the Node-B are generally assigned

higher modulation with higher code rates (e.g. 16-QAM and 3/4 code rate),

and both decreases as the distance between UE and Node-B increases.

The HSDPA-capable UE can support the use of5, 10 and 15 multi-codes. When

a UE receives 15 multi-codes with a 16-QAM modulation scheme and no cod-

ing (effective code rate of one), the maximum peak data rate it can experience

is 14.4 Mbits.

Rel. 5 defines twelve new categories for HSDPA UEs according to the fol-

lowing parameters (see Figure 1.6):

- Maximum number of HS-DSCH multi-codes that the UE can simultane-

ously receive (5, 10 or 15).

- Minimum inter-TTI time, which defines the minimum time between the

beginning of two consecutive transmissions to that UE. An inter-TTI of

one means that the UE can receive HS-DSCH packets during consecutive

TTIs (i.e. every 2 ms); an inter-TTI of two means that the scheduler would

need to skip one TTI between consecutive transmissions to that UE.

- Maximum number of HS-DSCH transport block bits received within an

HS-DSCH TTI. The combination of this parameter and the inter-TTI in-

terval determines the UE peak data rate.

- The maximum number of soft channel bits over all the HARQ processes.

A UE with a low number of soft channel bits will not be able to support

Incremental Redundancy (cf. sec. 1.3.2) for the highest peak data rates

and its performance will thus be slightly lower than for a UE supporting

a larger number of soft channels.

- Supported modulations (QPSK only or both QPSK and 16-QAM).

AMC provides a link adaptation functionality at Node B is in charge of

adapting the modulation, the coding format, and the number of multi-codes to

the instantaneous radio conditions.


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1.3. New Features 9

UE categoryof codes

Minimum inter Transport ch. Total numberof soft bits

Modulation Max. peak

1 5 3 7300 19200 QPSK & 16-QAM 1.2 Mbps

2 5 3 7300 28800 QPSK & 16-QAM 1.2 Mbps

3 5 2 7300 28800 QPSK & 16-QAM 1.8 Mbps

4 5 2 7300 38400 QPSK & 16-QAM 1.8 Mbps

5 5 1 7300 57600 QPSK & 16-QAM 3.6 Mbps

6 5 1 7300 67200 QPSK & 16-QAM 3.6 Mbps

7 10 1 14600 115200 QPSK & 16-QAM 7.2 Mbps

8 10 1 14600 134400 QPSK & 16-QAM 7.2 Mbps

9 15 1 20432 172800 QPSK & 16-QAM 10.2 Mbps

10 15 1 28776 172800 QPSK & 16-QAM 14.4 Mbps

11 5 2 3650 14400 QPSK only 0.9 Mbps

12 5 1 3650 QPSK only 1.8 Mbps

TTI interval

Max. number

data ratebits per TTI

Figure 1.6: HSDPA UE categories

1.3.2 Fast Hybrid Automatic Repeat reQuest

HSDPA uses HARQ (Hybrid Automatic Repeat Request) retransmission mecha-

nism with Stop and Wait (SAW) protocol. HARQ mechanism allows the UE

to rapidly request retransmission of erroneous transport blocks until they are

successfully received. HARQ functionality is implemented atMAC-hs (Medium

Access Control - high speed) layer, which is a new sub-layer for HSDPA. MAC-hs

is terminated at node B, instead of RLC (Radio Link Control) which is termi-

nated at RNC (Radio Network Controller). This involves a shorter retransmis-

sion delay (< 10 ms) for HSDPA than Rel. 99 (up to 100 ms). In order to better

use the waiting between acknowledgments, multiple processes can run for the

same UE using separate TTIs. This is referred to as N-channel SAW (N = up

to six for Advanced Node B implementation). In this way, while a channel

is waiting for an acknowledgment, the remaining N 1 channels continue to

transmit. HSDPA support both Chase Combining (CC) [6] and Incremental Re-

dundancy (IR).

CC consists in the retransmission from Node B of the same set of coded

symbols of the original packet. The decoder at the receiver combines these

multiple copies of the transmitted packet weighted by the received SNR prior

to decoding (see Figure 1.8). This type of combining provides time diversity

and soft combining gain at a low complexity cost and imposes the least de-

manding UE memory requirements of all Hybrid ARQ strategies. The com-


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bination process incurs a minor combining loss to be around 0.2 0.3 dB per

retransmission [7].

The state diagram of Figure 1.7(a) summarizes how the Chase Combining al-

gorithm works.

DataBlock

ErrorDetection

AcceptData Block

Transmission

Blocki

nErro

r

Retransmit the

NoErro

rNo Error

Deliver to Upper Layer

Block in Error

ErrorDetection

HARQ

same block

(a) Chase Combining

DataBlock

ErrorDetection

AcceptData Block

OriginalTransmission

Blocki

nErro

r

transmission

NoErro

rNo Error

Deliver to Upper Layer

Block in Error

ErrorDetection

HARQ

New

(b) Incremental Redundancy

Figure 1.7: IR and CC state diagrams

IR, on the other hand, sends different redundancy information during the

re-transmissions (see Figure 1.9). This leads to an incremental increasing of the

coding gain that can result in fewer retransmissions than for CC. IR is then par-

ticularly useful when the initial transmission uses high coding rates (e.g. 3/4)

but it implies higher memory requirements for the mobile receivers and larger

amount of control signaling compared to Chase Combining. Incremental Re-

dundancy can be further classified in Partial IR and Full IR. Partial IR includes

the systematic bits in every coded word, which implies that every retransmis-

sion is self-decodable, whereas Full IR only includes parity bits, and therefore

its retransmissions are not self-decodable. According to [7], Full IR only pro-

vides a significant coding gain for effective coding rates higher than 0.4 0.5,

because for lower coding rates the additional coding rate is negligible, since

the coding scheme is based on a 1/3 coding structure. On the other hand, for

higher effective coding rates the coding gain can be significant, for example a

coding rate of0.8 provides around 2 dB gain in Vehicular A (3km/h) scenario

with a QPSK modulation.

The state diagram of Figure 1.7(b) summarizes how the Chase Combining al-

gorithm works.

For a performance comparison of HARQ with Chase Combining and Incre-


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1.3. New Features 11

mental Redundancy for HSDPA systems see [8].

Effective rate after

R = 1

R = 1/2

R = 1/3

Original Transmission

1 Retransmission

2 Retransmission

st

nd

+

+

Datasoft combining

Figure 1.8: An example of Chase Combining retransmission

Original Data

Coded bits. Rate = 1/3

Effective rate after soft combiningat decoder stage

R = 1

R = 1/2

R = 1/3

Original Transmission

1 Retransmission

2 Retransmission

st

nd

+

+

Figure 1.9: An example of Incremental Redundancy retransmission

1.3.3 Fast Scheduling

The scheduler is a fundamental element of HSDPA, it affects its behavior and

its performance. At each TTI, the scheduler determines toward which terminal

(or terminals) the HS-DSCH should transmit and, together with AMC, at which

data rate. The HSDPA scheduler is located at the Node B. The algorithms used

to schedule are Round Robin (RR), Maximum Carrier to Interference (Max C/I)

and Proportional Fair (PF).

RR schedules users with a first-in first-out approach. This approach in-

volves a high fairness among all users, but at the same time it produces

a reduction of the overall system throughput since users can be served

even when they are experiencing weak signal.


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Maximum C/I schedules only users that are experiencing the maximum

C/I during that TTI. This scheme provides the maximum throughput for

the system but it produces unfairness of treatment among users penaliz-

ing those located at cell edge.

PF offers a good trade-off between RR (high fairness and low throughput)

and Maximum C/I (low fairness and high throughput). PF schedules

users according the ratio between their instantaneous achievable data

rate and their average served data rate.

1.4 Comparative Study of HSDPA and WCDMA

We have described how HSDPA Rel. 5 represents an evolution of WCDMA

Rel. 99 consisting in the introduction of a high speed transport channel (HS-

DSCH) and three new features: fast scheduling, fast link adaptation and fast

Hybrid ARQ. The aim of these three new tools is that of providing a rapid

adaptation to changing radio conditions. To achieve this aim, their functional-

ities are placed at the node B instead of the RNC as for the WCDMA.

As depicted in Table 1.2 [9], some CDMA features have been changed in the

HSDPA. In particular, Table 1.2 shows how the CDMA fast power control has

been replaced by fast Adaptive Modulation and Coding (AMC) causing an HS-

DPA power efficiency gain due to the elimination of the power control over-

head. In addition, AMC provides a fast link adaptation, which is achieved

following the policy that better are the link conditions experienced by the ter-

minal, higher is the data with which they are served. Another change concerns

the Spreading Factor (SF): in the CDMA it varies between 4 and 256 while in

the HSDPA it assumes a fixed value of 16. To support different data rates, the

HSDPA supports a wide combination of channel coding rates and modulation

format while WCDMA implements only the combination TC=1/3 and QPSK.

In order to increase the AMC efficiency and the link adaptation rate, the packet

duration has been reduced from 10 or 20 ms (Rel. 99) to a fixed value of 2 ms

(Rel. 5). To decrease the round trip time (RTT), i.e. the round trip delay, the

MAC funtionality of HS-DSCH has been placed at the node-B instead that at

the RNC.


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1.5. EVOLUTION OF HSDPA 13

Another difference is about the retransmission functionality. The WCDMA

Rel. 99 implements a simple ARQ scheme (the retransmitted packet are identi-

cal to those of the first transmission) while HSDPA Rel. 5 implements an HARQ

which supports both Chase Combining (CC) [6] and Incremental Redundancy

(IR). The last difference is about the CRC policy: while in the WCDMA the

CRC is implemented for each transport block, in the HSDPA it is implemented

for each TTI (i.e. it uses a single CRC for all transport blocks in the TTI) with a

consequent decrease of the overhead.

Feature Rel.99 DSCH Rel.5 HS-DSCH

Variable spreading factor Yes (4 - 256) No (16)

Fast power control Yes (1500 Hz) No

Fast rate control No (QPSK, TC=1/3) Yes (AMC, 500 Hz)

Fast L1 HARQ No ( 100 ms) Yes ( 10 ms)

HARQ with soft combining No CC or IR

TTI 10 or 20 ms 2 ms

Location of Mac RNC Node-B

CRC attachment per transport block per TTI

Peak data rate 2 Mbps 10 Mbps

Table 1.2: Comparison between DSCH and HS-DSCH basic properties

1.5 Evolution of HSDPA

HSDPA is making impressive inroads in the commercial service arena. The

Global mobile Suppliers Association (GSA) survey HSDPA Operator Commitments

published on January 2, 2007 reports 140 HSDPA networks in various stages of

deployment in 64 countries, of which 93 have commercially launched in 51

countries [10]. It means that HSDPA is today delivering commercial mobile

broadband services in North and South America, throughout Europe (includ-

ing 24 of the 27 EU nations), Asia, Africa, the Middle East and Australia.


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HSUPA

Whereas HSDPA optimizes downlink performance, High Speed Uplink Packet

Access (HSUPA), which uses the Enhanced Dedicated Channel (E-DCH), consti-

tutes a set of improvements that optimize uplink performance. These improve-

ments include higher throughputs, reduced latency, and increased spectral ef-

ficiency. HSUPA is standardized in Release 6. HSUPA will result in an approx-

imately 85 percent increase in overall cell throughput on the uplink and an ap-

proximately 50 percent gain in user throughput. HSUPA also reduces packet

delays. Such an improved uplink will benefit users in a number of ways. For

instance, some user applications transmit large amounts of data from the mo-

bile station, such as sending video clips or large presentation files. For future

applications such as VoIP, improvements will balance the capacity of the up-

link with the capacity of the downlink. HSUPA achieves its performance gains

through the following approaches:

- An enhanced dedicated physical channel.

- A short TTI, as low as 2 ms, which allows faster responses to changing

radio conditions and error conditions.

- Fast Node-B-based scheduling, which allows the base station to efficiently

allocate radio resources.

- Fast Hybrid ARQ, which improves the efficiency of error processing.

The combination of TTI, fast scheduling, and Fast Hybrid ARQ also serves

to reduce latency, which can benefit many applications as much as improved

throughput. HSUPA can operate with or without HSDPA in the downlink,

though it is likely that most networks will use the two approaches together.

The improved uplink mechanisms also translate to better coverage, and for ru-

ral deployments, larger cell sizes. Apart from improving uplink performance,

E-UL improves HSDPA performance by making more room for acknowledg-

ment traffic and by reducing overall latency. HSUPA can achieve different

throughput rates based on various parameters, including the number of codes

used, the spreading factor of the codes, the TTI value, and the transport block

size in bytes, as illustrated in Figure 1.10.


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categoryCodes

Spreading TransportTTI

1 1 4 7296 10 0.73 Mbps

2 2 4 14592 10 1.46 Mbps

2 2 4 2919 2 1.46 Mbps

3 2 4 14592 10 1.46 Mbps

4 2 2 20000 10 2.00 Mbps

4 2 2 5837 2 2.90 Mbps

5 2 2 20000 10 2.00 Mbps

6 2 + 2 2 + 4 20000 10 2.00 Mbps

6 2 + 2 2 + 4 11520 2 5.76 Mbps

Factor

Data rateblock size

HSUPA

Figure 1.10: HSUPA peak throughput rates

The combination of HSDPA and HSUPA is called High-Speed Packet Ac-

cess (HSPA).

Evolution of HSPA (HSPA+)

Wireless and networking technologists are developing a continual series of en-

hancements for HSPA, some of which are being specified in Release 6 and Re-

lease 7, and some of which are being studied for Release 8. 3GPP has specified

a number of advanced receiver designs, including Type 1 which uses mobile

receive diversity, Type 2 which uses channel equalization and Type 3, which

includes a combination of receive diversity and channel equalization.

The first approach, specified in Rel. 6, is mobile-receive diversity. This

technique relies on the optimal combining of received signals from separate

receiving antennas. The antenna spacing yields signals that have somewhat

independent fading characteristics. Hence, the combined signal can be more

effectively decoded, which results in a downlink capacity gain of up to 50 per-

cent when employed in conjunction with techniques such as channel equal-

ization. Receive diversity is effective even for small devices such as PC Card

modems and smartphones. Current receiver architectures based on rake re-

ceivers are effective for speeds up to a few megabits per second. But at higher

speeds, the combination of reduced symbol period and multipath interference

results in inter-symbol interference and diminishes rake receiver performance.

This problem can be solved by advanced receiver architectures such as chan-

nel equalizers that yield an additional 20 percent gain over HSDPA with re-


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ceive diversity. Alternative advanced receiver approaches include interference

cancellation and generalized rake receivers (G-Rake). Different vendors are

emphasizing different approaches. However, the performance requirements

for advanced receiver architectures are specified in 3GPP Release 6. The com-

bination of mobile receive diversity and channel equalization (Type 3) is espe-

cially attractive as it results in a large gain independently of the radio channel.

What makes such enhancements attractive is that no changes are required to

the networks except increased capacity within the infrastructure to support

the higher bandwidth. Moreover, the network can support a combination of

devices, including both earlier devices that do not include these enhancementsand those that do. Device vendors can selectively apply these enhancements

to their higher performing devices.

Another capability being standardized is Multiple Input Multiple Output.

MIMO refers to a technique that employs multiple transmit antennas and mul-

tiple receive antennas, often in combination with multiple radios and multiple

parallel data streams. The most common use of the term MIMO applies to

spatial multiplexing. The transmitter sends different data streams over each

antenna. Whereas multipath is an impediment for other radio systems, MIMO

actually exploits multipath, relying on signals to travel across different com-

munications paths. This results in multiple data paths effectively operating

somewhat in parallel and, through appropriate decoding, in a multiplicative

gain in throughput. Tests of MIMO have proven very promising in WLANs

operating in relative isolation, where interference is not a dominant factor. Spa-

tial multiplexing MIMO should also benefit HSPA hotspots serving local ar-

eas such as airports, campuses, and malls, where the technology will increase

capacity and peak data rates. However, in a fully loaded network with inter-

ference from adjacent cells, overall capacity gains will be more modest, in the

range of 20 to 33 percent over mobile-receive diversity. Although MIMO can

significantly improve peak rates, other techniques such as Space Division Mul-

tiple Access (SDMA) (also a form of MIMO) may be even more effective than

MIMO for improving capacity in high spectral efficiency systems using a reuse

factor of 1. 3GPP has enhanced the system to support SDMA operation as part

of Rel. 6. In Rel. 7, Continuous Packet Connectivity enhancements reduce the


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uplink interference created by dedicated physical control channels of packet

data users when they have no user data to transmit. This helps increase the

limit for the number of HSUPA users that can stay connected at the same time.

3GPP currently has a study item referred to as HSPA Evolution or HSPA+

that is not yet in a formal specification development stage. The intent is to cre-

ate a highly optimized version of HSPA that employs both Rel. 7 features and

other incremental features such as interference cancellation and optimizations

to reduce latency.

The goals of HSPA+ are to:

- Exploit the full potential of a CDMA approach before moving to an OFDM

platform in 3GPP LTE.

- Achieve performance comparable to Long Term Evolution (LTE) in 5 MHz

of spectrum.

- Provide smooth interworking between HSPA+ and LTE that facilitates

operation of both technologies. As such, operators may choose to lever-

age the SAE planned for LTE.

- Allow operation in a packet-only mode for both voice and data.

- Be backward compatible with previous systems while incurring no per-

formance degradation with either earlier or newer devices.

- Facilitate migration from current HSPA infrastructure to HSPA+ infras-

tructure.

3GPP Long Term Evolution (LTE)

Although HSPA and HSPA+ offer a highly efficient broadband wireless ser-

vice that will likely enjoy success for the remainder of the decade, 3GPP is also

working on a project called Long Term Evolution. LTE will allow operators

to achieve even higher peak throughputs in higher spectrum bandwidth. Ini-

tial possible deployment is targeted for 2009. LTE uses Orthogonal Frequency

Division Multiple Access (OFDMA) on the downlink, which is well suited to

achieve high peak data rates in high spectrum bandwidth. WCDMA radio


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technology is about as efficient as OFDM for delivering peak data rates of

about 10 Mbps in 5 MHz of bandwidth. However, achieving peak rates in

the 100 Mbps range with wider radio channels would result in highly complex

terminals and is not practical with current technology. It is here that OFDM

provides a practical implementation advantage. Scheduling approaches in the

frequency domain can also minimize interference, and hence boost spectral ef-

ficiency. On the uplink, however, a pure OFDMA approach results in high Peak

to Average Ratio (PAR) of the signal, which compromises power efficiency and

ultimately battery life. Hence, LTE uses an approach called SC-FDMA, which

has some similarities with OFDMA but will have a 2 to 6 dB PAR advantageover the OFDMA method used by other technologies such as IEEE 802.16e.

LTE goals include:

- Downlink peak data rates up to 100 Mbps with 20 MHz bandwidth.

- Uplink peak data rates up to 50 Mbps with 20 MHz bandwidth.

- Operation in both TDD and FDD modes.

- Scalable bandwidth up to 20 MHz, covering 1.25 MHz, 2.5 MHz, 5 MHz,10 MHz, 15 MHz, and 20 MHz in the study phase. 1.6 MHz wide chan-

nels are under consideration for the unpaired frequency band, where a

TDD approach will be used.

- Increase spectral efficiency over Rel. 6 HSPA by a factor of two to four.

- Reduce latency to 10 ms round-trip time between user equipment and

the base station and to less than 100 ms transition time from inactive to

active.

The overall intent is to provide for an extremely high-performance radio-

access technology that offers full vehicular speed mobility and that can readily

coexist with HSPA and earlier networks. Because of scalable bandwidth, oper-

ators will be able to easily migrate their networks and users from HSPA to LTE

over time.

The impressive improvements in the achievable peak data rates due to LTE

will lead, in the next years, to the spreading of rich multimedia services and


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applications over wireless networks. Since these services require the using of

TCP (Transmission Control Protocol), TCP issues, performance, and enhancing

solutions over HSDPA networks will be extensively discussed in Chapter 2 and

in Chapter 3.


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Chapter 2

TCP Overview

2.1 TCP Architecture

The distinctive characteristic of 3rd Generation wireless networks is packet

data services. The information provided by these services are, in the major-

ity of the cases, accessible on the Internet. Since internet communications are

for the almost entirety constituted by TCP traffic, the research community is

showing a wide interest in extending TCP application in mobile and wireless

networks.

TCP is a connection oriented transport protocol which provides a reliable

byte stream to the application layer [11]. Reliability is achieved using ARQ

mechanism based on positive acknowledgments. TCP provides transparent

segmentation and reassembly of user data and handles flow and congestion

control. TCP packets are cumulatively acknowledged when they arrive in se-

quence, out of sequence packets cause the generation of duplicate acknowledg-

ments. TCP manages a retransmission timer which is started when a segment

is transmitted. Retransmission timers are continuously updated on a weighted

average of previous round trip time (RTT) measurements, i.e. the time it takes

from the transmission of a segment until the acknowledgment is received. TCP

sender detects a loss either when multiple duplicate acknowledgments (the de-

fault value is 3) arrive, implying that the next packet was lost, or when a re-

transmission timeout (RTO) expires. The RTO value is calculated dynamically

21


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22 CHAPTER 2. TCP Overview

based on RTT measurements. This explains why accuracy in RTT measure-

ments is critical: delayed timeouts slow down recovery, while early ones may

lead to redundant retransmissions.

A prime concern for TCP is congestion. Today all TCP implementations are

required to use algorithms for congestion control, namely slow start, conges-

tion avoidance, fast retransmit and fast recovery [12].

Slow Start and Congestion Avoidance

Since TCP was initially designed to be used in wired networks where transmis-

sion losses are extremely low (BERs in the order of 1010, and down to 1012

for optical links), TCP assumes that all losses are due to congestion. Therefore,

when TCP detects packet losses, it reacts both retransmitting the lost packet

and reducing the transmission rate. In this way it allows router queues to

drain. Afterwards, it gradually increases the transmission rate to probe the

networks capacity.

The purpose of slow start and congestion avoidance is to prevent the conges-

tion from occurring varying the transmission rate. TCP maintains a congestion

window (cwnd), which represents an estimate of the number of segments that

can be injected into the network without causing congestion (a segment is any

TCP data or acknowledgment packet (or both)). The initial value of the con-

gestion window is between one and four segments [13]. The receiver maintains

an advertised windows (rwnd) which indicates the maximum number of bytes it

can accept. The value of the rwnd is sent back to the sender together with each

segment going back. At any moment, the amount of outstanding data (wnd)

is limited by the minimum of cwnd and rwnd, i.e. new packets are only sent

if allowed by both congestion window and receivers advertised window, assummarized by

wnd = min(rwnd,cwnd) (2.1)

In the slow start phase, the congestion window is increased by one segment

for each acknowledgment received (cwnd = cwnd+1). This phase is used both

when new connections are established and after retransmissions due to time-


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2.1. TCP Architecture 23

outs occurring. The slow start phase causes an exponential increase of the

congestion window and it lasts until a timeout occurs or a threshold value

(ssthresh) is reached. When the cwnd reaches the ssthresh value, the slow start

phase ends and the congestion avoidance phase starts. While the slow start al-

gorithm opens the congestion window quickly to reach the limit capacity of the

link as rapid as possible, the congestion avoidance algorithm is conceived to

transmit at a safe operating point and increase the congestion window slowly

to probe the network for more bandwidth becoming available.

In the congestion avoidance phase, the congestion window is increased by one

packet per round trip time, which gives a linear increase of the window. More

precisely, for each non duplicate ACK received the cwnd is increased according

to the following equation:

cwnd = cwnd + MSSMSS/cwnd (2.2)

Equation (2.2) provides an acceptable approximation to the underlying prin-

ciple of increasing cwnd by 1 full-sized segment per RTT [12].

When a timeout occurs, the ssthresh is reduced to one-half the current win-

dow size (equation (2.3)), the congestion window is reduced to one MSS (Max-

imum Segment Size), and the slow start phase in entered again.

ssthresh =min

(rwnd,cwnd)/2 (2.3)

Figure 2.1 shows an example of how the congestion window changes dur-

ing the slow start and the congestion avoidance phase. In this example the

initial ssthresh is set to 16 and a timeout occurs after 8 round trip times. At that

time, the cwnd assumes a value of 20, hence the new threshold after timeout

(new sstresh) is set to 10.


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Con

gestio

nWind

ow

(segmen

ts)

Round trip times

0

2

4

6

10

12

14

16

18

ssthresh

20

8

8 9 13121110 141 2 3 4 5 6 70

new ssthresh

Timeout

Figure 2.1: TCP slow start and congestion avoidance phase

Fast Retransmit and Fast Recovery

The fast retransmit and fast recovery algorithms allow TCP to detect data loss

before the transmission timer expires. These algorithms permit to increase TCP

performance in two ways: allowing earlier loss detection and retransmission

and not reducing the transmission rate as much as after timeout.

When an out of order segment arrives, the receiver transmits an acknowl-

edgment referred to the segment it was expected to receive. The purpose of

this duplicate acknowledgment (dupack) is to inform the sender that a segment

was received out of order, and to tell it what sequence number is expected. The

fast retransmit and fast recovery algorithms are usually implemented togetheras follows.

After receiving three dupacks in a row, the sender concludes that the miss-

ing segment was lost. Therefore, TCP performs a direct retransmission of the

missing segment after the reception of the third dupack (the fourth acknowl-

edgment) even if the retransmission timer has not expired. The ssthresh is set

to the same value as in the case of timeout (equation (2.3)). After the retrans-

mission, fast recovery is performed until all lost data is recovered. The con-


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2.1. TCP Architecture 25

gestion window is set to three segments more than ssthresh. These additional

three segments take account of the number of segments (three) that have left

the network and which the receiver has buffered. For each additional dupli-

cate acknowledgment received, the cwnd is incremented by one (cwnd=cwnd+1)

as well as in slow start phase, since each dupack indicates that one segment

has left the network. The fast recovery phase ends when a non-duplicate ac-

knowledgment arrives. The congestion window is then set to the same value as

ssthresh and it is incremented by one segment for RTT as well as in congestion

avoidance phase (equation (2.2)).

With fast retransmit and fast recovery, TCP is able to avoid unnecessary

slow starts due to minor congestion incidents (dupacks are indicators of some

kind of network congestion but it is not as strict as a timeout).

Cong

estio

nWin

dow

(segmen

ts)

Round trip times

0

2

4

6

10

12

14

16

18

ssthresh

20

8

8 9 13121110 141 2 3 4 5 6 70

new ssthresh

3 duplicate ACK(Fast retransmitting)

rd

Fast recovery

New ACK

Figure 2.2: TCP fast retransmit and fast recovery phase


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2.2 TCP Problems over 3G Networks

TCP has been designed for wired networks where packet losses are almost

negligible and where packet losses and delays are mainly caused by conges-

tion. Instead, in wireless networks the main source of packet losses is the link

level error of the radio channel, which may seriously degrade the achievable

throughput of the TCP protocol. Thus, TCP performance over wireless net-

works can differ from TCP performance over wired networks.

The main problem with TCP performance in networks that have both wired

and wireless links is that packet losses that occur because of bad channel con-

ditions are mistaken by the TCP sender as being due to network congestion,

causing it to drop its transmission window, resulting in degraded throughput.

From a wireless performance point of view, the flow control represents one of

the most important aspects of TCP. The flow control is in charge of determin-

ing the load offered by the sender to achieve maximum connection throughput

while preventing network congestion or receivers buffer overflow.

The main characteristics of wireless networks that can affect TCPs perfor-

mance are the following:

Block Error Rate

As mentioned above, in wired networks losses are mainly due to conges-

tion caused by buffer overflows. Wireless networks are instead charac-

terized by high bit error rate (BER). If these errors are not corrected, they

lead to block error rate (BLER). Since TCP flow and congestion control

mechanisms assume that losses are only due to congestion, when packet

losses due to corruption in the wireless link occur, TCP congestion control

mechanism will react reducing the cwnd and resetting the retransmission

timer. This TCP erroneous interpretation of errors leads to poor perfor-

mance due to under utilization of the bandwidth and to very high delay

jitter.

Latency

Latency in 3G wireless networks is mainly due to transmission delays in

the radio access network and to the extensive processing required at the

physical layer. Larger latency can be mistaken for congestion.


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2.2. TCP Problems over 3G Networks 27

Delay spikes

A delay spike is a sudden increase in the latency of the link [14] . The

main causes of delay spikes are:

- Link layer recovery from a outage due to a temporal loss of radio

coverage (e.g. driving into a tunnel)

- Inter-frequency handovers or inter-system handovers. Inter-frequen-

cy handovers occur when the UE is handed over another operators

Node B that uses different frequency; inter-system handovers occur

passing from a technologies to another (e.g. from 2G to 3G).

- High priority traffic (e.g. voice) can block low priority applications

(e.g. data connection) whether terminals do not handle both voice

and data connection at the same time. In this case, low priority ap-

plications can be suspended so that high priority ones can be com-

pleted.

Delay spikes can cause spurious TCP timeouts (cf. sec. 3.3), unnecessary

retransmissions and a multiplicative decrease in the cwnd size.

Serial Timeouts

When the connection is paused for a certain time (for example, due to

hard-handover), several retransmissions of the same segment can be lost

during this pause. Since TCP uses an exponential backoff mechanism,

when a timeout occurs TCP increases the retransmission timeout by some

factor (usually, a doubling) before retransmitting the unacknowledged

data. This increasing lasts until the RTO reaches a limit value (usually,

about a minute). This means that when mobile resumes its connection,

there is the possibility that no data will be transmitted for up to a minute,degrading the performance drastically.

Data Rates

Data rates in wireless networks are very dynamic due to mobility, vary-

ing channel conditions, effects from other users and even from varying

demands from the connection. Moreover, when user move into another

cell he can experience a sudden change in available data rate. An increas-

ing in the available bandwidth can lead to an under utilization of it due


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to TCP slow start phase. On the other hand, when the data rate decrease,

the TCP congestion control mechanism takes care of it but sudden RTT

increase can cause a spurious TCP timeout [14].

2.3 TCP Versions

In this section some different congestion control and avoidance mechanisms

will be studied, which have been proposed for TCP/IP protocols, namely:

Tahoe, Reno, NewReno, Westwood Vegas, SACKand FACK.

Each of the above implementations suggest a different mechanism to deter-

mine when a segment should be retransmitted and how should the sender be-

have when it encounters congestion. In addition, they suggest what pattern of

transmission they have to follow to avoid congestion.

TCP Tahoe

TCP Tahoe refers to the TCP congestion control algorithm proposed in [15].

This implementation adds new algorithms and refinements to earlier imple-

mentations. The new algorithm include slow-start, congestion avoidance and fast

retransmit (cf. sec. 2.1). The refinements include a modification to the round

trip time estimator used to set retransmission timeout values.

The problem of Tahoe is that it takes a complete timeout interval to detect a

packet loss. In addition, it performs slow start if a packet loss is detected even

if some packet can still flow through the network. This leads to an abrupt re-

ducing of the flow.

TCP Reno

TCP Reno retains the enhancements incorporated into Tahoe adding to the fast

recovery phase the fast recovery algorithm [16].

TCP Reno provides an important enhancement compared to TCP Tahoe, pre-

venting the communication path (usually called pipe) from going empty after

fast retransmit, thereby avoiding the need to slow start to re-fill it after a single

packet loss.


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2.3. TCP VERSIONS 29

Renos fast recovery mechanism is optimized for the case when a single packet

is dropped from a window of data but it can suffer from performance problems

when multiple packets are dropped from a window of data. In the case of mul-

tiple packets dropped, Renos performance are almost the same as Tahoe. This

is due to the fact that the fast recovery algorithm mechanism implemented by

TCP Reno can lead to a stall. Indeed, TCP Reno goes out of fast recovery when

it receives a new partial ACK (i.e. a new ACK which does not represent an

ACK for all outstanding data). That means that if a lot of segments from the

same window are lost, TCP Reno is pulled out of fast recovery too soon, and it

may stall since no new packets can be sent.

TCP NewReno

NewReno [17] represents a slight modification over TCP Reno. It is able to de-

tect multiple packet losses and thus it appears much more efficient than TCP

Reno when they occur. NewReno, as well as Reno, enters into fast retransmit

when it receives multiple duplicate packets, but differently from the latter it

does not exit from fast recovery phase until all outstanding data at the time it

entered fast recovery are acknowledged. This means that in NewReno partial

ACK do not take TCP out of fast recovery but they are treated as an indica-

tor that the packet immediately following the acknowledged packet in the se-

quence space has been lost, and should be retransmitted. Thus, when multiple

packets are lost from a single window of data, NewReno can recover without a

retransmission timeout, retransmitting one lost packet per round trip time un-

til all of the lost packets from that window have been retransmitted. NewReno

exits fast recovery phase when all data outstanding when this phase was initi-

ated has been acknowledged (i.e., it exits fast recovery when all data injected

into network, and still waiting for an acknowledgment at the moment that fast

recovery was initiated, has acknowledged).

The main NewRenos issue is that it takes one round trip time to detect each

packet loss.


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TCP Westwood

TCP Westwood represents a modified version of TCP Reno since it enhances

the window control and backoff process [18].

Westwood sender monitors the acknowledgment stream it receives and from

it estimates the data rate currently achieved by the connection. Whenever the

sender perceives a packet loss (i.e. a timeout occurs or 3 DUPACKs are re-

ceived), the sender uses the bandwidth estimate to properly set the congestion

window and the slow start threshold. By backing off to cwnd and ssthresh val-

ues that are based on the estimated available bandwidth (rather than simply

halving the current values as Reno does), TCP Westwood avoids reductions of

cwnd and ssthresh that can be excessive or insufficient. In this way TCP West-

wood ensures both faster recovery and more effective congestion avoidance.

Experimental studies reveal the benefits of the intelligent backoff strategy in

TCP Westwood: better throughput, goodput and delay performance.

TCP SACK

TCP with Selective Acknowledgment represents an extension of TCP Reno and

NewReno. It provides a solution both to the problem of the detection of mul-

tiple lost packets and to the retransmission of more than one lost packet per

round trip time.

TCP SACK requires that segments are acknowledged selectively rather than

cumulatively. It uses the option field in the TCP header to store a set of prop-

erly received sequence numbers [19].

During fast recovery, SACK maintains a variable called pipe, that represents

the estimated number of packets outstanding on the link. The sender only

sends new or retransmitted data when the value ofpipe is less than the cwnd.

The variable pipe is incremented each time the sender sends a packet, and is

decremented when the sender receives duplicate ACK with a SACK option re-

porting that new data has been correctly received. When the sender is allowed

to send a packet, it sends the next packet known as missing at the receiver if

such a packet exists, otherwise it sends a new packet. When a retransmitted

packet is lost, SACK detects it through a classic RTO and then goes into slow


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2.3. TCP VERSIONS 31

start. The sender only goes out of fast recovery when an ACK is received ac-

knowledging all data that was outstanding when fast recovery was entered.

Because of this, SACK appears closer to NewReno than to Reno, since partial

ACKs do not pull the sender out of fast recovery.

TCP FACK

TCP with Forward Acknowledgment is an extension of TCP SACK. It has the

same functionalities of TCP SACK but it introduces some improvements com-

pared to it:

- A more precise estimation of outstanding. It uses SACK option to better

estimate the amount of data in transit [20].

- A data smoothing. It introduces a better way to halve the window when

congestion is detected. When the cwnd is immediately halved, the sender

stops transmitting for a while and then resumes when enough data has

left the network. This unequal distribution of segments over one RTT can

be avoided when the window is gradually decreased [20].

- A new slow start and congestion control. When congestion occur, the

window should be halved according to the multiplicative decrease of the

correct cwnd. Since the sender identifies congestion at least one RTT after

it happened, if during that RTT it was in Slow Start mode, then the cur-

rent cwnd will be almost double than the cwnd when congestion occurred.

Therefore, in this case, the cwnd is first halved to estimate the correct cwnd

that should be further decreased.

TCP Vegas

In contrast to the TCP Reno algorithm which induces congestion to learn the

available network capacity, Vegas algorithm anticipates the onset of congestion

by monitoring the difference between the rate it is expecting to see and the rate

it is actually realizing [21]. Vegas strategy is to adjust the sources sending rate

(i.e. the cwnd) in an attempt to keep a small number of packets buffered in the

routers along the transmission path. The TCP Vegas sender stores the current


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value of the system clock for each segment it sends. By doing so, it is able to

know the exact RTT for each sent packet.

The main innovations introduced by TCP Vegas are the following:

- New retransmission mechanism. When a duplicate acknowledgment is re-

ceived, the sender checks if(current time - segment transmission time)>RTT.

If it is true, the sender provides a retransmission without waiting for the

classic retransmission timeout nor for three duplicate ACKs. To catch

any other segments that may have been lost prior to the retransmission,

when a duplicate acknowledgment is received, if it is the first or second

one after a fresh acknowledgment then it again checks the timeout val-

ues and if the segment time exceeds the timeout value then it retransmits

the segment without waiting for a duplicate ACK. In this way Vegas can

detect multiple packet losses.

Moreover, it only reduces its window if the retransmitted segment was

sent after the last decrease. Thus it also overcome Renos shortcoming

of reducing the congestion window multiple time when multiple packets

are lost.

- New congestion control mechanism. TCP Vegas does not use segment losses

to signal that there is congestion. It determines congestion by calculating

the difference between the calculated throughput and the value it would

achieve if the network was not congested. If that difference is smaller

than a boundary, the window is increased linearly to make use of the

available bandwidth, otherwise it is decreased linearly to prevent over

saturating the bandwidth. The throughput of an uncongested network is

defined as the window size in bytes divided by the BaseRTT, which is the

value of the RTT in an uncongested network.

- New slow start mechanism. The cwnd is doubled every time the RTT chang-

es instead of every RTT. The reason for this modification is that when a

connection starts for the first time the sender has no idea of the available

bandwidth. Thus it may happen that during exponential increase it over

shoots the available bandwidth by a big amount inducing congestion.


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2.4. ROUND TRI P TIME AND MEAN NUMBER OF RETRANSMISSIONS FOR TCPOVER 3G 33

The slow start phase in terminated when a boundary value is reached in

the difference between the current RTT and the last RTT. This represents

a modification compared to others TCP versions where the boundary is

set in the cwnd size.

TCP

Tahoe

TCP

Reno

TCP

N.Reno

TCP

West.

TCP

SACK

TCP

FACK

TCP

Vegas

Slow Start Yes Yes Yes Yes Yes Enhanc.

Version

Enhanc.

Version

CongestionAvoidance

Yes Yes Yes Yes Yes Yes Enhanc.Version

Fast Retransmit Yes Yes Yes Yes Yes Yes Yes

Fast Recovery No Yes Enhanc.

Version

Enhanc.

Version

Enhanc.

Version

Enhanc.

Version

Yes

Retransmission

mechanism

Normal Normal Normal Normal Normal Normal New

mechan.

Congestion Con-

trol mechanism

Normal Normal Normal Normal Normal New

mechan.

New

mechan.

Selective ACKmechanism

No No No No Yes Yes No

Table 2.1: TCP versions comparison

2.4 Round Trip Time and mean number of retrans-

missions for TCP over 3G

A correct estimate of the round trip time is fundamental. The round trip time

represents a merit figure of any connection since it gives an indication on how

fast the transmitter can react to any event that occurs in the connection. It could

be defined as the elapsed period since the transmitter sends a packet until it re-

ceives the corresponding acknowledgement. With the purpose of accelerating

such transmitter response time, the round trip time should be minimized as

much as possible.


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In HSDPA, the size of a TCP segment is 1500 byte and each TTIs lasts 2 ms.

According to the modulation and coding schemes used on the radio interface,

transmitting a TCP segment requires since 12 up to 60 TTIs. How well known,

the wireless channel presents variable characteristics both from the point of

view of link conditions (expressed in terms of block error rate (BER)) and from

that of transmission time delay.

Let [22] NTTI(i) the number of transmissions of TTI i due to HARQ, Tj the

transmission time of a segment on the radio interface (it depends by the bit rate

chosen by the the scheduler), RTTwired the average RTT of the wired part of

the network, ns the number of TTIs needed to transmit a TCP segment whenno errors occurs on the radio interface. Then the round trip time (RTT) of the

whole link (wired part plus wireless part) is given by:

RTT =

ns

i=1NTTI(i)

nsTj + RTTwired (2.4)

The term:

Ni =ns

i=1

NTTI(i)ns

(2.5)

represents the number of transmissions of a TCP segment (Ni). Since errors

on each TTI are independent and identical distributed (i.i.d.) [23], Ni can be

modelled by a Gaussian variable. Then, also the RTT expressed by equation 2.4

can be modelled by a Gaussian variable. Is now possible to define the mean Ns

[23] [22] [24] [25] and the variance 2 [23] [24] ofNi:

Ns =1 + Pe PePs

1 PePs(2.6)

2 =Pe(1 Pe + PePs)

(1 PePs)2(2.7)

where Ps is the probability of errors after soft combining two successive trans-

mission of the same information block and Pe is the probability of errors after


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2.4. ROUND TRI P TIME AND MEAN NUMBER OF RETRANSMISSIONS FOR TCPOVER 3G 35

decoding the information block, i.e. it represents the BLER. In such way, we

have defined Ni N(Ns, 2).

From Figure 2.3 and Figure 2.4 we can extract Ns and 2 values corresponding

to different values of BLER.

Figure 2.3: Mean value Ns as a function of BLER [26]

Figure 2.4: Variance 2 as a function of BLER [26]


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Chapter 3

TCP Enhancing Solutions

The proposals to optimize TCP for wireless links can be divided into three cat-

egories: link layer, end-to-end and split connection solutions.

Link layer solutions try to reduce the error rate of the link through some kind of

retransmission mechanism. As the data rate of the wireless link increase, there

will be more time for multiple link level retransmissions before timeout occurs

at the TCP layer, making link layer solutions more viable. In sec. 3.4 a link layer

solution named Snoop will be analysed.

End-to-end solutions try to modify the TCP implementation at the sender and/or

receiver and/or intermediate routers, or optimizing the parameters used by the

TCP connection to achieve good performance. The end-to-end solution named

Eifel will be analysed in sec. 3.3

Split connections try to separate the TCP used in the wireless link from the one

used in the wired one. The optimization procedure can there be done sepa-

rately on the wired and wireless part. A proxy solution will be analysed in

sec. 3.1.

The solutions proposed in sections 3.1, 3.3 and 3.4 will be then utilized dur-

ing the simulations of Chapter 4.

3.1 Proxy Solution

Proxy solutions consist in splitting the connection between the sender (i.e. the

server) and the terminal (i.e. the UE) by means of an interposed proxy. This

37


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38 CHAPTER 3. TCP Enhancing Solutions

solution permits to split the connection serverterminal into one connection

between the server and the proxy, and another between the proxy and the ter-

minal (see Figure 3.1). In this way, the server will continue to see an ordinary

wired network while changing in the system will be made only to the proxy

and possibly to the terminal. This solution has been introduced by [27] and it

is also known with the name split TCP.

RNCNode B ServerTerminal

TCP

(a)

ProxyRNCNode B ServerTerminal

TCP TCP

(b)

Figure 3.1: Proxy solution architecture

An accurate studying about proxy solution over WCDMA networks is

downlink tcp proxy solutions over hsdpa with multiple data flow

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