Downlink TCP Proxy Solutions over HSDPA with Multiple Data Flow DANIELE GIRELLA

Transcription

1 Downlink TCP Proxy Solutions over HSDPA with Multiple Data Flow DANIELE GIRELLA Master s Degree Project Stockholm, Sweden 2007

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3 Downlink TCP Proxy Solutions over HSDPA with Multiple Data Flow DANIELE GIRELLA Master s Degree Project February 2007 Automatic Control Group School of Electrical Engineering

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5 Abstract In recent years, several proxy solutions have been proposed to improve performance 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 successful 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 user s 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 simulator. 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 functionality 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 enhancing protocols (such as Eifel). The simulator allows for performance comparison of existing TCP solution over wireless, and the proxy we have studied in this thesis. An analysis of the effects of multi-user data flows over TCP performance have been also addressed. iii

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7 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 gamingthat 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 Downlink 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 of 3 rd Generation wireless networks is packet data services. The information provided by these services are, in the majority 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 wireless networks is that it has been designed for wired networks where packet losses are almost negligible and where delays are mainly caused by congestion. 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 v

8 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 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. 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 wireless 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 implementation called EURANE, a comparative study of all the above solutions in an HSDPA scenario will be provided.

9 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 provided 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 italian 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 Speranzon for helping me when I was in trouble with some control systems and to Niels Möller for helping me with ns2. Now is the moment to thank all those that have left a mark on my life during my five years of studying at University of L Aquila. 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 vii

10 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 my happiness, and made me happy. Thanks for the love, patience, understanding, and for putting your unreserved confidence in me. Thanks for being a so special 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, endless 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.

11 Contents 1 High Speed Downlink Packet Access (HSDPA) HSDPA Concept Channel Structure New Features Adaptive Modulation and Coding Fast Hybrid Automatic Repeat request Fast Scheduling Comparative Study of HSDPA and WCDMA Evolution of HSDPA TCP Overview TCP Architecture TCP Problems over 3G Networks TCP Versions Round Trip Time and mean number of retransmissions for TCP over 3G TCP Enhancing Solutions Proxy Solution Flow Aggregation Eifel Protocol Snoop Protocol Further Enhancing Protocols ix

12 x CONTENTS 4 Simulation ns-2 Simulator and EURANE extension Simulation Scenario Simulation Results Conclusions 73 References 75

13 List of Figures 1.1 HS-PDSCH channel time and code multiplexing HS-SCCH frame structure HS-DPCCH frame structure [1] HSDPA channel functionality HSDPA physical layer HSDPA UE categories IR and CC state diagrams An example of Chase Combining retransmission An example of Incremental Redundancy retransmission HSUPA peak throughput rates TCP slow start and congestion avoidance phase TCP fast retransmit and fast recovery phase Mean value N s as a function of BLER [26] Variance σ 2 as a function of BLER [26] Proxy solution architecture RNF signalling TCP flow aggregation scheme [30] Sample logical aggregate for a give Mobile Host [30] Eifel procedure [35] Snoop procedure [39] UE side MAC architecture [50] UTRAN side overall MAC architecture [50] Main characteristics of EURANE s schedulers [52] xi

14 xii LIST OF FIGURES 4.4 Overview of physical layer model used in EURANE [52] Simulation scenario Available link bandwidth Network architecture UE s throughput in the simple scenario Server s congestion window in simple and RNFProxy scenarios Trends obtained in simple scenario setting server s cwnd to Throughput improvements by adding Eifel and Snoop protocols UE s throughput in RNFProxy scenario Throughput improvements by adding Eifel and Snoop protocols to RNFProxy scenario UE s throughput in RNFProxy scenario with both Eifel and Snoop protocols Comparison between throughput s trend in simple scenario and in RNFProxy scenario Comparison between throughput s trend in RNFProxy scenario (with and without enhancing protocols) Comparison between the throughput experienced interposing a RNFProxy and that experienced with a SimpleProxy

15 List of Tables 1.1 2G to 3G throughput comparison Comparison between DSCH and HS-DSCH basic properties TCP versions comparison Scenario s characteristics Simulation parameters Summary of simulation results xiii

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17 List of Abbreviations 3G 3GPP ACB ACK AMC ARQ ATCP BCH BDP BER BLER C/I CC CDMA CQI CRC CWND DCH DPCH DSCH DUPACK E-DCH EDGE EURANE Third Generation 3 rd Generation Partnership Project Aggregate Control Block Acknowledgment Adaptive Modulation and Coding Automatic Repeat Request Aggregate TCP Broadcast Channel Bandwidth Delay Product Bit Error Rate Block Error Rate Carrier to Interference Ratio Chase Combining Code Division Multiple Access Channel Quality Indicator Cyclic Redundancy Check Congestion window Dedicated Channel Dedicated Physical Channel Downlink Shared Channel Duplicate Acknowledgment Enhanced Dedicated Channel Enhanced Data rates for Global Evolution Enhanced UMTS Radio Access Network Extension xv

18 xvi LIST OF ABBREVIATIONS FACH FACK FDD FH GGSN GPRS GSM H-ARQ HSDPA HS-DPCCH HS-DSCH HS-PDSCH HS-SCCH HSPA HSUPA IR LTE MAC MAC-b MAC-c MAC-d MAC-hs MAC-sh MCS MH MIMO MSR MSS OFDM OFDMA PCH PF QAM Forward Access Channel Forward Acknowledgment Frequency Division Duplex Fixed Host Gateway GPRS Support Node General Packet Radio Service Global System for Mobile communication Hybrid Automatic Repeat Request High-Speed Downlink Packet Access High-Speed Dedicated Physical Control Channel High-speed Downlink Shared Channel High-Speed Physical Downlink Shared Channel High-Speed Shared Control Channel High-Speed Packet Access High Speed Uplink Packet Access Incremental Redundancy Long Term Evolution Medium Access Control Medium Access Control for BCH Medium Access Control for PCH Medium Access Control for DCH Medium Access Control high-speed Medium Access Control for DSCH Modulation and Coding Scheme Mobile Host Multiple Input Multiple Output Mobile Support Router Maximum Segment Size Orthogonal Frequency Division Multiplexing Orthogonal Frequency Division Multiplexing Access Paging Channel Proportional Fair Quadrature amplitude modulation

19 LIST OF ABBREVIATIONS xvii QPSK RACH RLC RNC RNF RR RTO RTT RWND SACK SAW SDMA SF SGSN SH SIR SSTHRESH SYN TCP TDD TLE TTI UDP UE UMTS UTRAN VOIP WCDMA WLAN Quadrature Phase Shift Keying Random Access Channel Radio Link Control Radio Network Controller Radio Network Feedback Round Robin Retransmission Timeout Round Trip Time Receiver Window Selective Acknowledgment Stop And Wait Space Division Multiple Access Spreading Factor Serving GPRS Support Node Supervisory Host Signal to Interference Ratio Slow Start Threshold Synchronize Transmission Control Protocol Time Division Duplex Transmission Layer Efficiency Transmission Time Interval User Datagram Protocol User Equipment Universal Mobile Telecommunication System UMTS Terrestrial Radio Access Network Voice over IP Wideband Code Division Multiple Access Wireless Local Area Network

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21 Chapter 1 High Speed Downlink Packet Access (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-fivefold sector throughput increase, which means more data users on a single frequency (or carrier). The impressive increase of data rate is achieved by implementing a fast and complex channel control mechanism based upon short physical layer frames (cf. sec. 1.2), Adaptive Modulation and Coding (AMC) (cf. sec ), fast Hybrid-Automatic Repeat request (H-ARQ) (cf. sec ) and fast scheduling (cf. sec ). 1

22 2 CHAPTER 1. High Speed Downlink Packet Access (HSDPA) GSM GPRS EDGE UMTS HSDPA Typical max. data rate Theoret. peak data rate 9.6 kbps 40 kbps 120 kbps 384 kbps Mbps 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 incremental 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

23 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) consists of 15 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) of 16 is used with a maximum of 15 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). HSDPA frame = 2 ms Standard Rel. 99 frame = 10 ms Spreading Codes user 1 user 2 user 3 user 4 user 5 Figure 1.1: HS-PDSCH channel time and code multiplexing In order to support the HS-DSCH operation, an HSDPA UE needs new control channels: the HS-SCCH in the downlink direction and the HS-DPCCH in

24 4 CHAPTER 1. High Speed Downlink Packet Access (HSDPA) the uplink direction. HS-SCCH is a fixed rate (60 Kbps, SF=128) channel used for carrying downlink 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 changing parameters that are needed for HS-DSCH reception. This includes HARQrelated information and the parameters of the HS-DSCH transport format selected by the link adaptation mechanism (see Figure 1.2). T = 0.67 ms slot modulation scheme, code set, UE id 2 x T slot= 1.33 ms 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.

25 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

26 Uplink Downlink 6 CHAPTER 1. High Speed Downlink Packet Access (HSDPA) DL associated DPCH (for each HSDPA user) Slot (0.67 ms) HS-SCCH HS-PDSCH #1 HS-PDSCH #2 TTI (2 ms) HS-PDSCH # slots HS-DPCCH CQI CQI ACK CQI UL associated DPCH (for each HSDPA user) 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 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 denominated CQI that provides implicit information about the instantaneous signal quality received by the user. The CQI specifies the modulation, the number of codes and the transport block size the UE can support

27 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 commands 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 instantaneous status of the user s 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 detector (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 user s 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 conditions. 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 combinations of modulation and channel coding-rate can be used to provide different

28 8 CHAPTER 1. High Speed Downlink Packet Access (HSDPA) 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 of 5, 10 and 15 multi-codes. When a UE receives 15 multi-codes with a 16-QAM modulation scheme and no coding (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 following parameters (see Figure 1.6): - Maximum number of HS-DSCH multi-codes that the UE can simultaneously 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 interval 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 ) 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.

29 1.3. New Features 9 UE category Max. number Minimum inter Transport ch. Total number Modulation Max. peak of codes TTI interval bits per TTI of soft bits data rate QPSK & 16-QAM 1.2 Mbps QPSK & 16-QAM 1.2 Mbps QPSK & 16-QAM 1.8 Mbps QPSK & 16-QAM 1.8 Mbps QPSK & 16-QAM 3.6 Mbps QPSK & 16-QAM 3.6 Mbps QPSK & 16-QAM 7.2 Mbps QPSK & 16-QAM 7.2 Mbps QPSK & 16-QAM 10.2 Mbps QPSK & 16-QAM 14.4 Mbps QPSK only 0.9 Mbps QPSK only 1.8 Mbps Figure 1.6: HSDPA UE categories Fast Hybrid Automatic Repeat request HSDPA uses HARQ (Hybrid Automatic Repeat Request) retransmission mechanism 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 at MAC-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 terminated at RNC (Radio Network Controller). This involves a shorter retransmission 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 Redundancy (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 demanding UE memory requirements of all Hybrid ARQ strategies. The com-

30 10 CHAPTER 1. High Speed Downlink Packet Access (HSDPA) bination process incurs a minor combining loss to be around db per retransmission [7]. The state diagram of Figure 1.7(a) summarizes how the Chase Combining algorithm works. Data Block Error Detection Transmission Block in Error Retransmit the same block No Error HARQ Error Detection No Error Accept Data Block Block in Error Deliver to Upper Layer Data Block Original Transmission Error Detection New transmission Block in Error No Error HARQ Error Detection No Error Accept Data Block Block in Error Deliver to Upper Layer (a) Chase Combining (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 particularly 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 Redundancy can be further classified in Partial IR and Full IR. Partial IR includes the systematic bits in every coded word, which implies that every retransmission is self-decodable, whereas Full IR only includes parity bits, and therefore its retransmissions are not self-decodable. According to [7], Full IR only provides a significant coding gain for effective coding rates higher than , 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 of 0.8 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 algorithm works. For a performance comparison of HARQ with Chase Combining and Incre-

31 1.3. New Features 11 mental Redundancy for HSDPA systems see [8]. Data Original Transmission Effective rate after soft combining R = 1 st 1 Retransmission + R = 1/2 nd 2 Retransmission + R = 1/3 Figure 1.8: An example of Chase Combining retransmission Original Data Coded bits. Rate = 1/3 Effective rate after soft combining at decoder stage Original Transmission R = 1 st 1 Retransmission + R = 1/2 nd 2 Retransmission + R = 1/3 Figure 1.9: An example of Incremental Redundancy retransmission 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 involves 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.

32 12 CHAPTER 1. High Speed Downlink Packet Access (HSDPA) 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 penalizing 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 functionalities 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 overhead. In addition, AMC provides a fast link adaptation, which is achieved following the policy that better are the link conditions experienced by the terminal, 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.

33 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 identical 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 (including 24 of the 27 EU nations), Asia, Africa, the Middle East and Australia.

34 14 CHAPTER 1. High Speed Downlink Packet Access (HSDPA) HSUPA Whereas HSDPA optimizes downlink performance, High Speed Uplink Packet Access (HSUPA), which uses the Enhanced Dedicated Channel (E-DCH), constitutes a set of improvements that optimize uplink performance. These improvements include higher throughputs, reduced latency, and increased spectral efficiency. HSUPA is standardized in Release 6. HSUPA will result in an approximately 85 percent increase in overall cell throughput on the uplink and an approximately 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 mobile station, such as sending video clips or large presentation files. For future applications such as VoIP, improvements will balance the capacity of the uplink 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 rural deployments, larger cell sizes. Apart from improving uplink performance, E-UL improves HSDPA performance by making more room for acknowledgment 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.

35 1.5. EVOLUTION OF HSDPA 15 HSUPA Spreading Transport Codes category Factor block size TTI Data rate Mbps Mbps Mbps Mbps Mbps Mbps Mbps Mbps Mbps Figure 1.10: HSUPA peak throughput rates The combination of HSDPA and HSUPA is called High-Speed Packet Access (HSPA). Evolution of HSPA (HSPA+) Wireless and networking technologists are developing a continual series of enhancements for HSPA, some of which are being specified in Release 6 and Release 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 percent when employed in conjunction with techniques such as channel equalization. Receive diversity is effective even for small devices such as PC Card modems and smartphones. Current receiver architectures based on rake receivers 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 channel equalizers that yield an additional 20 percent gain over HSDPA with re-

36 16 CHAPTER 1. High Speed Downlink Packet Access (HSDPA) 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 combination of mobile receive diversity and channel equalization (Type 3) is especially 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 enhancements and 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 multiple 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 communications 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. Spatial multiplexing MIMO should also benefit HSPA hotspots serving local areas such as airports, campuses, and malls, where the technology will increase capacity and peak data rates. However, in a fully loaded network with interference 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 Multiple 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

37 1.5. EVOLUTION OF HSDPA 17 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 create 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 leverage 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 performance degradation with either earlier or newer devices. - Facilitate migration from current HSPA infrastructure to HSPA+ infrastructure. 3GPP Long Term Evolution (LTE) Although HSPA and HSPA+ offer a highly efficient broadband wireless service 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. Initial possible deployment is targeted for 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

38 18 CHAPTER 1. High Speed Downlink Packet Access (HSDPA) 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 efficiency. 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 advantage over the OFDMA method used by other technologies such as IEEE e. 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 channels 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 radioaccess technology that offers full vehicular speed mobility and that can readily coexist with HSPA and earlier networks. Because of scalable bandwidth, operators 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

39 1.5. EVOLUTION OF HSDPA 19 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.

40

41 Chapter 2 TCP Overview 2.1 TCP Architecture The distinctive characteristic of 3 rd Generation wireless networks is packet data services. The information provided by these services are, in the majority 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 sequence, out of sequence packets cause the generation of duplicate acknowledgments. 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 default value is 3) arrive, implying that the next packet was lost, or when a retransmission timeout (RTO) expires. The RTO value is calculated dynamically 21

42 22 CHAPTER 2. TCP Overview based on RTT measurements. This explains why accuracy in RTT measurements 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, congestion avoidance, fast retransmit and fast recovery [12]. Slow Start and Congestion Avoidance Since TCP was initially designed to be used in wired networks where transmission losses are extremely low (BERs in the order of 10 10, and down to 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 network s capacity. The purpose of slow start and congestion avoidance is to prevent the congestion 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 congestion 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 receiver s advertised window, as summarized 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-

43 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 algorithm 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 + MSS MSS/cwnd (2.2) Equation (2.2) provides an acceptable approximation to the underlying principle of increasing cwnd by 1 full-sized segment per RTT [12]. When a timeout occurs, the ssthresh is reduced to one-half the current window size (equation (2.3)), the congestion window is reduced to one MSS (Maximum 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 during 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.

44 24 CHAPTER 2. TCP Overview 20 Timeout 18 Congestion Window (segments) ssthresh new ssthresh Round trip times 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 acknowledgment 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 together as follows. After receiving three dupacks in a row, the sender concludes that the missing segment was lost. Therefore, TCP performs a direct retransmission of the missing segment after the reception of the third dupack (the fourth acknowledgment) 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 retransmission, fast recovery is performed until all lost data is recovered. The con-

45 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 duplicate 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 acknowledgment 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). 20 rd 3 duplicate ACK (Fast retransmitting) 18 Congestion Window (segments) ssthresh Fast recovery new ssthresh New ACK Round trip times Figure 2.2: TCP fast retransmit and fast recovery phase

46 26 CHAPTER 2. TCP Overview 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 congestion. 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 networks 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 conditions 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 determining the load offered by the sender to achieve maximum connection throughput while preventing network congestion or receiver s buffer overflow. The main characteristics of wireless networks that can affect TCP s performance are the following: Block Error Rate As mentioned above, in wired networks losses are mainly due to congestion caused by buffer overflows. Wireless networks are instead characterized 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 performance 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.

47 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-frequency 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 applications can be suspended so that high priority ones can be completed. 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, varying 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 increasing in the available bandwidth can lead to an under utilization of it due

48 28 CHAPTER 2. TCP Overview 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, SACK and FACK. Each of the above implementations suggest a different mechanism to determine when a segment should be retransmitted and how should the sender behave 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 implementations. 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 reducing 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, preventing 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.

49 2.3. TCP VERSIONS 29 Reno s 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 multiple packets dropped, Reno s 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 detect 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 indicator that the packet immediately following the acknowledged packet in the sequence 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 until all of the lost packets from that window have been retransmitted. NewReno exits fast recovery phase when all data outstanding when this phase was initiated 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 NewReno s issue is that it takes one round trip time to detect each packet loss.

50 30 CHAPTER 2. TCP Overview 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 received), the sender uses the bandwidth estimate to properly set the congestion window and the slow start threshold. By backing off to cwnd and ssthresh values 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 Westwood 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 multiple 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 properly 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 of pipe 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 reporting 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

51 2.3. TCP VERSIONS 31 start. The sender only goes out of fast recovery when an ACK is received acknowledging 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 compared 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 current 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 source s 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

52 32 CHAPTER 2. TCP Overview 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 received, 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 values 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 Reno s 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 changes 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.

53 2.4. ROUND TRIP TIME AND MEAN NUMBER OF RETRANSMISSIONS FOR TCP OVER 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 TCP TCP TCP TCP TCP TCP Tahoe Reno N.Reno West. SACK FACK Vegas Slow Start Yes Yes Yes Yes Yes Enhanc. Version Enhanc. Version Congestion Avoidance Yes Yes Yes Yes Yes Yes Enhanc. Version Fast Retransmit Yes Yes Yes Yes Yes Yes Yes Fast Recovery No Yes Enhanc. Enhanc. Enhanc. Enhanc. Yes Version Version Version Version Retransmission mechanism Normal Normal Normal Normal Normal Normal New mechan. Congestion Control mechanism Normal Normal Normal Normal Normal New mechan. New mechan. Selective mechanism ACK No No No No Yes Yes No Table 2.1: TCP versions comparison 2.4 Round Trip Time and mean number of retransmissions 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 receives the corresponding acknowledgement. With the purpose of accelerating such transmitter response time, the round trip time should be minimized as much as possible.

54 34 CHAPTER 2. TCP Overview 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] N T T I (i) the number of transmissions of TTI i due to HARQ, T j the transmission time of a segment on the radio interface (it depends by the bit rate chosen by the the scheduler), RT T wired the average RTT of the wired part of the network, n s the number of TTIs needed to transmit a TCP segment when no errors occurs on the radio interface. Then the round trip time (RTT) of the whole link (wired part plus wireless part) is given by: RT T = ns i=1 N T T I(i) n s T j + RT T wired (2.4) The term: N i = n s i=1 N T T I (i) n s (2.5) represents the number of transmissions of a TCP segment (N i ). Since errors on each TTI are independent and identical distributed (i.i.d.) [23], N i 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 N s [23] [22] [24] [25] and the variance σ 2 [23] [24] of N i : N s = 1 + P e P e P s 1 P e P s (2.6) σ 2 = P e(1 P e + P e P s ) (1 P e P s ) 2 (2.7) where P s is the probability of errors after soft combining two successive transmission of the same information block and P e is the probability of errors after

55 2.4. ROUND TRIP TIME AND MEAN NUMBER OF RETRANSMISSIONS FOR TCP OVER 3G 35 decoding the information block, i.e. it represents the BLER. In such way, we have defined N i N(N s, σ 2 ). From Figure 2.3 and Figure 2.4 we can extract N s and σ 2 values corresponding to different values of BLER. Figure 2.3: Mean value N s as a function of BLER [26] Figure 2.4: Variance σ 2 as a function of BLER [26]

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57 Chapter 3 TCP Enhancing Solutions The proposals to optimize TCP for wireless links can be divided into three categories: 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 separately on the wired and wireless part. A proxy solution will be analysed in sec The solutions proposed in sections 3.1, 3.3 and 3.4 will be then utilized during the simulations of Chapter 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

58 38 CHAPTER 3. TCP Enhancing Solutions solution permits to split the connection server terminal into one connection between the server and the proxy, and another between the proxy and the terminal (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. Terminal Node B TCP RNC Server (a) Terminal Node B TCP RNC Proxy TCP Server (b) Figure 3.1: Proxy solution architecture An accurate studying about proxy solution over WCDMA networks is reported in [28], where it is shown how local knowledge (in the proxy) about the state of a TCP connection can be used to enhance performance by shortcutting the ACKs transmission or packets retransmission. Moreover, it demonstrates that split TCP solution is particularly useful for radio links with high data rates, since they are characterized by a large bandwidth delay product. The proxy solution used in this thesis is the one proposed by [29], which allows to improve both the user experience of wireless internet and the utilization of the existing infrastructure. The proxy-based scheme introduced in [29] uses a new custom protocol between the RNC and the proxy. This protocol provides information from the data-link layer within the RNC to the transport layer within the proxy. This

59 3.2. Flow Aggregation 39 communication is called Radio Network Feedback (RNF) and it is sent via UDP (User Datagram Protocol). The RNF message is sent from the RNC to the proxy every time the available link bandwidth over the wireless channel is computed. The link bandwidth represents the instantaneous channel capacity of the wireless link, computed with a given frequency. When the proxy receives the RNF message, it takes appropriate action by adjusting the TCP window size. The computation of the cwnd in the proxy also takes into consideration the queue in the RNC. It is important to note that bandwidth variations act as a disturbance which is possible to measure but not to affect, while the queue length is a parameter that is possible to affect. This is the reason why the part of RNF message concerning the available bandwidth is a feed-forward while the part concerning the queue length is a feedback. Figure 3.2 shows how the RNF signalling works. Variable BW RNC RNF message RNFProxy UE Node B queue - recompute cwnd - update cwnd Control sender s rate Server Figure 3.2: RNF signalling 3.2 Flow Aggregation In conventional TCP implementations every connection is independent and thus for each is kept a different state information (such as cwnd, ssthresh and so on). However, since all TCP connections to a mobile host share the same wireless link, they are statistically dependant thus flows to the same mobile host might share certain TCP state information. The solution proposed in [30] treats all the flows to the same mobile host as a single aggregate. The scheme is depicted in Figure 3.3. Treating all TCP flows to a particular mobile host as an