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Quality of service enhancements in IEEE 802.11 wireless LANs Wong, George Wai 2003

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Quality of Service Enhancements in IEEE 802.11 Wireless LANs by George Wai Wong B.A.Sc, The University of British Columbia, Canada, 2001 A THESIS S U B M I T T E D IN P A R T I A L F U L F I L L M E N T O F T H E REQUIREMENTS FOR T H E DEGREE OF M A S T E R O F APPLIED S C I E N C E in T H E F A C U L T Y O F G R A D U A T E STUDIES D E P A R T M E N T O F E L E C T R I C A L AND C O M P U T E R E N G I N E E R I N G  We accept this thesis as conforming to the required standard  T H E UNIVERSITY O F BRITISH C O L U M B I A September 2003 © George Wai Wong, 2003  In presenting this thesis in partial fulfilment of the requirements for an advanced degree at the University of British Columbia, I agree that the Library shall make it freely available for reference and study. I further agree that permission for extensive copying of this thesis for scholarly purposes may be granted by the head of my department or by his or her representatives. It is understood that copying or publication of this thesis for financial gain shall not be allowed without my written permission.  Department of Electrical and Computer Engineering The University of British Columbia Vancouver, Canada  Date: September 26 , 2003.  Abstract One of the most important of the many developments in telecommunications is the convergence o f voice, video, and data communications within the Internet Protocol (IP) suite. IP was originally designed to support best-effort data services. The development o f IP-based multimedia networking applications has imposed Quality of Service (QoS) requirements on the IP network, especially with respect to real-time traffic.  The Internet Engineering Task  Force (IETF) is currently working on Q o S differentiation at the LP layer; however the result is sub-optimal without lower layers' support.  W i t h the increasing use of wireless Internet  services over 802.11 wireless L A N s , it is essential to focus on Q o S differentiation support at the 802.11 M A C layer.  To improve the current 802.11 M A C protocol, the I E E E Task Group E was formed and is defining QoS enhancements for the 802.11 M A C layer by introducing an Enhanced Distributed Coordination Function ( E D C F ) . The E D C F provides prioritization enhancement of the 802.11 Distributed Coordination Function ( D C F ) .  The objective of this thesis is to propose and evaluate a novel packet retransmission algorithm called Age-Dependent Backoff to improve the Q o S performance of E D C F .  The  A D B algorithm dynamically varies the persistent factor associated with the contention window, based on real-time packet queue age and lifetime.  A D B maintains the backward  compatibility property o f E D C F and involves relatively easy implementation at low cost.  Extensive simulation results obtained using O P N E T software show that E D C F with the A D B retransmission algorithm provides l o w values for delay, jitter, and drop rate for realtime traffic without sacrificing the throughput o f best-effort data traffic. novel and low cost means to improve the QoS performance of E D C F .  ii  A D B is a viable,  Table of Contents  Abstract  ii  Table o f Contents  iii  List of Tables  vi  List of Figures  vii  Acknowledgments  ix  Chapter 1.  Introduction  1  1.1  Background on I E E E 802.11 Wireless L A N s  1  1.2  Motivations  3  1.3  Thesis Contributions  4  1.4  Thesis Outline  5  Chapter 2.  Overview of IEEE 802.11 Wireless LANs  7  2.1  Introduction to the I E E E 802.11 Standard  7  2.1.1  802.11 Topology  8  2.1.2  802.11 Protocol Architecture  10  2.2  802.11 D C F Protocol  12  2.3  N e w 802.1 l e E D C F Protocol  14  Chapter 3.  Age-Dependent Backoff Algorithm for 802.1 le EDCF  18  3.1  Backoff Behaviour o f 802.11e E D C F  18  3.2  Age-Dependent Backoff Algorithm  19  3.3  Implementation Complexity  21  Implementation of the A D B Algorithm  21  3.3.1  iii  3.3.2  Discussion o f the Implementation Complexity  24  Chapter 4.  Design of Simulation Models  25  4.1  Overview o f O P N E T Simulation Tools  26  4.2  802.11 M o d e l Design  28  4.2.1  802.11 Physical Layer M o d e l  28  4.2.2  802.11 M A C Layer M o d e l  30  Simulation Models for Upper Layer Protocols  36  4.3.1  Network Layer M o d e l : Internet Protocol  36  4.3.2  Transport Layer Models: T C P and U D P  37  Traffic Source Models  40  4.4.1  Voice Model  41  4.4.2  Video Conferencing M o d e l  42  4.4.3  F T P File Transfer M o d e l  43  4.3  4.4  Chapter 5  in an Ad-Hoc Network and a Hotspot  45  A d - H o c Network Scenario  46  5.1.1  Overview of the A d - H o c Network Scenario  46  5.1.2  Simulation Results from the A d - H o c Network Scenario  47  5.1.3  Discussions o f the A d - H o c Network Results  51  Hotspot Scenario  52  5.2.1  Overview o f the Hotspot Scenario  52  5.2.2  Simulation results from the Hotspot Scenario  54  5.2.3  Discussions o f the Hotspot Results Conclusion from the Simulation Results  61 62  5.1  5.2  5.3  Performance Analysis of the Proposed ADB Algorithm  iv  Chapter 6.  Summary and Conclusions  64  6.1  Summary of the Work  65  6.2  Future Work  66  References  67  v  List of Tables  Table 4.1.  Parameters for the M A C layer  35  Table 4.2.  Default settings o f the T C P parameters  39  Table 4.3.  Parameters for video conferencing  43  Table 4.4.  Parameters for F T P file transfers  44  vi  List of Figures  Figure 2.1.  I E E E reference model comparing with the OSI reference model  8  Figure 2.2.  Independent basic service set  9  Figure 2.3.  Extended service set  9  Figure 2.4.  802.11 protocol layers  10  Figure 2.5.  802.11 D C F M A C protocol operation  12  Figure 2.6.  R T S / C T S / d a t a / A C K and N A V settings  14  Figure 2.7.  Service differentiation by different A I F S values  15  Figure 2.8.  Transmission architecture of E D C F vs. D C F  16  Figure 3.1.  P F [ T C ] vs. A g e  20  Figure 3.2.  F l o w chart for the implementation of the A D B retransmission algorithm.. 23  Figure 4.1.  Hierarchical structure for O P N E T models  Figure 4.2.  Finite state machine for 802.11 M A C layer model with E D C F and A D B . . 33  Figure 4.3.  Two-state model of a voice user  41  Figure 5.1.  Network topology of an ad-hoc scenario  47  Figure 5.2.  V o i c e packet delay in an ad-hoc network  48  Figure 5.3.  V o i c e packet jitter in an ad-hoc network  48  Figure 5.4.  V o i c e packet drop rate in an ad-hoc network  49  Figure 5.5.  Video packet delay in an ad-hoc network  49  Figure 5.6.  Video packet jitter in an ad-hoc network  50  Figure 5.7.  Video packet drop rate in an ad-hoc network  50  Figure 5.8  Total F T P data throughput in an ad-hoc network  51  vii  26  Figure 5.9.  Network topology o f a hotspot scenario  53  Figure 5.10.  Uplink and Downlink voice packet delay in a hotspot  55  Figure 5.11a. Uplink voice packet j itter in a hotspot  56  Figure 5.11b. Downlink voice packet jitter in a hotspot  56  Figure 5.12a. Uplink voice packet drop rate in a hotspot  57  Figure 5.12b. Downlink voice packet drop rate in a hotspot  57  Figure 5.13.  58  Video packet delay in a hotspot  Figure 5.14a. U p l i n k video packet j itter in a hotspot  59  Figure 5.14b. Downlink video packet jitter in a hotspot  59  Figure 5.15a. U p l i n k video packet drop rate in a hotspot  60  Figure 5.15b. Downlink video packet drop rate in a hotspot  60  Figure 5.16.  61  Total F T P data throughput in a hotspot  viii  Acknowledgments  I wish to thank most sincerely my supervisor, Dr. Robert W . Donaldson, for his inexhaustible patience, encouragement and valuable assistance i n connection with this thesis. I would like to express my thanks to my colleague, Victor Fong, for participating in technical and non-technical discussions with me. I would also like to thank O P N E T technologies, Inc. for providing us a free academic version o f O P N E T Modeler. Finally, I am most grateful to m y parents not only for their support but also for their constant encouragement.  ix  Chapter 1 Introduction In this introductory chapter, background material on I E E E 802.11 wireless L A N s is reviewed i n section 1.1. The motivations for this thesis work are presented i n section 1.2. The thesis contributions are summarized in section 1.3. Finally, the outline o f the thesis is provided i n section 1.4.  1.1  Background on I E E E 802.11 Wireless LANs Wireless L A N products first appeared iri the late 1980s, marketed as substitutes for  traditional wireline L A N s [1].  The idea was to use wireless L A N s to avoid the cost o f  installing L A N cables and to ease the task o f relocating computer stations. The history o f wireless L A N development has been fraught with proprietary non-standard technologies which cause wireless L A N products being non-interoperable among vendors.  W i t h the  ratification o f the I E E E 802.11 wireless L A N standard, wireless L A N s have emerged from being proprietary implementations to become open solutions for providing mobility as well as essential network services where wireline installations are impractical [2].  The original I E E E 802.11 standard [3], published i n 1997, supports data rates up to 2 M b p s i n the 2.4-GHz industrial, scientific, and medical (ISM) band.  In 1999, The I E E E  802.11 Working Group published its enhanced versions, I E E E 802.11a [4] and 802.11b [5], that extend the data rate to 54 M b p s i n the 5 - G H z unlicensed national  information  infrastructure (UNII) band and 11 M b p s i n the 2.4-GHz I S M band, respectively. Recently, the I E E E has approved the final specification for I E E E 802.1 l g [6], which is backwards  1  compatible with 802.11b and which boosts the bandwidth capability to 54 Mbps in the 2.4G H z I S M band. Although most of the growth in 802.11 wireless L A N s has been around the 802.11b standard, pundits predict that the emergence of the 802.1 l g standard w i l l dominate the 802.11 wireless L A N market because of the full backward compatibility with 802.1 l b and the high data rate of 54 M b p s [7].  Despite not being finalized until recently, many vendors  have already been shipping 802.1 l g products based on the drafts of the 802.1 l g standard since late 2002.  Following the approval of 802.1 l g , the formation of I E E E 802.11 Task  Group N began. The data rate of 802.1 In has not been determined yet but is expected to be at least 108 Mbps or possibly beyond up to 320 Mbps.  The I E E E 802.11 wireless L A N standard is based on the definition of the medium access control ( M A C ) protocol and the physical layer ( P H Y ) specifications. The P H Y defines frequency bands, data rates, and other details of the actual wireless transmission. Above the P H Y is the M A C layer, which regulates access to the shared wireless channel so that station transmissions do not interfere with one another. The 802.11 M A C layer was designed to be common among different 802.11 P H Y s such as 802.11a, 802.11b and 802.1 l g . The 802.11 M A C layer protocol is specified in terms of coordination functions that determine when a wireless station is allowed to transmit data over the wireless medium.  The mandatory  Distributed Coordination Function ( D C F ) uses a carrier-sense multiple access/collision avoidance ( C S M A / C A ) protocol for sharing the wireless medium. This C S M A / C A protocol reduces the probability of collisions among stations which share the medium by using a random backoff time when the medium is busy. Once the medium is idle, a random backoff time defers a station from transmitting a frame, minimizing the chance o f inter-station collisions.  2  802.11 wireless LANs are today's most deployed wireless LANs and are expected to play a major role in next-generation wireless communications. The increasing use of wireless Internet services has created a strong demand for public Internet access over wireless LANs. It is envisioned that 802.11 will soon become one of the most common wireless Internet access technologies.  1.2 Motivations The world of communications has undergone many changes over the last few years. One  of the most important changes is the convergence  of voice, video, and data  communications within the Internet Protocol (IP) suite. IP was originally designed to support data services such as file transfer, e-mail and remote terminal, which are tolerant of delay and jitter. Voice and video services, as opposed to data services, require a minimum transmission rate and suffer significantly from high delay and jitter.  The development of IP-based  multimedia networking applications has imposed additional requirements on the IP network, creating a need for end-to-end Quality of Service (QoS) support.  Although the Internet  Engineering Task Force (IETF) is currently working on service differentiation at the IP layer, the result is sub-optimal without lower layers' support.  In recent years, there has been a  substantial increase in the deployment of 802.11 wireless LANs for wireless Internet services. Wireless Internet Service Providers (WISPs) are appearing everywhere, deploying 802.11 hotspots in coffee shops, hotels, and airports. With the growing popularity and acceptance of 802.11 wireless LANs, it is essential to focus on service differentiation support at the 802.11 M A C layer.  3  To improve the current 802.11 M A C protocol to support applications with QoS requirements, the I E E E 802.11 Task Group E was formed and is defining Q o S enhancements for the 802.11 M A C protocol.  The 802.1 l e draft introduces Enhanced Distributed  Coordination Function ( E D C F ) and Hybrid Coordination Function ( H C F ) , both of which are currently under discussion [8]. E D C F is a prioritization enhancement of the 802.11 D C F using the Virtual D C F mechanism [9].  E D C F is the contention-based channel access mechanism  for H C F which is based on a polling mechanism similar to the 802.11 Point Coordination Function ( P C F ) [10]. H C F allows the Hybrid Coordinator ( H C ) , typically located at the access point ( A P ) , to initial contention-free periods at any given time during a contention period.  The success of 802.11 wireless L A N s is based on D C F , which is a distributed protocol with minimum management and maintenance costs. The dynamic nature of ad-hoc networks makes it difficult to use a polling protocol relying on centralized control to maintain connection, reservation, and scheduling states.  W i t h these observations, we argue that a  distributed control of wireless channel results in a more productive use o f wireless resources than a centralized control.  Therefore, we w i l l focus on the improvement of E D C F in this  thesis.  1.3  Thesis Contributions The thesis extends the current work on providing Q o S enhancements in 802.11  wireless L A N s studied by I E E E 802.11 Task Group E.  In particular, we address the  improvement of the new 802.1 l e E D C F by proposing an efficient retransmission algorithm called Age-Dependent Backoff ( A D B ) [11], which dynamically adjusts the persistent factors  4  based on the ages o f the real-time packets i n the transmission queues and the lifetimes o f the real-time packets.  The complexity o f implementing the A D B algorithm is relatively low.  A D B requires minor modifications i n the computation o f C W minimizing the migration effort from the 802.1 l e E D C F and provides backward compatibility to the 802.11 D C F .  The performance o f E D C F with the A D B retransmission algorithm is evaluated using the O P N E T simulation tools. We test the A D B algorithm i n two typical environments, an adhoc network and a hotspot, i n which 802.11 wireless L A N s might most probably be applied. The ad-hoc network is used mostly i n situations where users need to deploy a network to start communication quickly without the benefit o f a fixed network irifrastructure.  The hotspot is  usually deployed by a W I S P , providing wireless Internet access services for public use. In both environments, voice, video and data services are active simultaneously. T w o simulation scenarios are implemented using O P N E T to model these two common environments. W e study the improvements i n service differentiation under the environments when the A D B retransmission algorithm is employed in the new 802.1 l e E D C F protocol.  The simulation results indicate that using A D B in E D C F provides l o w values for delay, jitter and drop rate for real-time traffic without sacrificing the throughput o f the besteffort traffic i n a wide range o f traffic loads and network configurations.  1.4  Thesis Outline The subsequent chapters o f this thesis are organized as follows.  In Chapter 2, we  review the 802.11 standard and introduce the new 802.1 l e E D C F protocol proposed by the 802.11 Task Group E. In Chapter 3, we describe our proposed A D B retransmission algorithm  5  for E D C F and discuss the complexity of A D B . The simulation model is presented in Chapter 4 and the performance analysis of 802.1 l e E D C F with A D B is studied i n Chapter 5. Finally, a summary and conclusions are provided in Chapter 6.  6  Chapter 2 Overview of IEEE 802.11 Wireless LANs Before studying the details o f anything, it often helps to get a general "lay o f the hand." A basic introduction is often necessary when studying networking topics because the number o f acronyms can be overwhelming. The 802.11 standard uses a significant number o f acronyms, which makes the introduction to be important.  This chapter introduces the  acronyms used throughout the entire thesis and provides an overview o f the I E E E 802.11 wireless L A N standard. The introduction to the 802.11 wireless L A N standard appears i n Section 2.1. The 802.11 D C F , a mandatory 802.11 M A C protocol, mentioned briefly i n the previous chapter is presented thoroughly i n Section 2.2. Finally, Section 2.3 describes the new 802.1 l e E D C F protocol proposed by the 802.11 Task Group E for supporting service differentiation at the M A C layer o f 802.11 wireless L A N s .  2.1  Introduction to the I E E E 802.11 Standard The 802.11 wireless L A N standard, officially called " I E E E Standard for Wireless  L A N M e d i u m Access Control ( M A C ) and Physical Layer ( P H Y ) Specifications," defines over-the-air protocols necessary to support networking i n a local area. A s with other I E E E 802-based standards (such as 802.3 Ethernet and 802.5 Token Ring), the primary service o f the 802.11 standard is to deliver M A C Service Data Units ( M S D U s ) between peer Logical L i n k Controls ( L L C s ) . A s a refresher for readers, the I E E E Standards Committee subdivided the Data L i n k layer i n the open systems interconnection (OSI) reference model developed by the International Organization for Standardization (ISO).  The result o f this subdivision  depicted i n Figure 2.1 split the Data L i n k layer into a M A C layer and a L L C layer [12]. L L C  7  is the highest layer o f the I E E E 802 reference model, providing addressing and data link control, and is independent o f the network topology, the transmission medium, and the M A C protocol. The 802.11 standard specifies M A C protocols and P H Y specifications for wireless connectivity o f fixed, portable, and mobile stations moving at pedestrian or vehicular speed within a local area. Wireless cards and access points typically provide functions o f the 802.11 standard.  OSI Reference Model  IEEE 802 Reference Model  Data Link Layer  Logical Link Layer (LLC) Medium Access Control (MAC)  Physical Layer  Physical Layer  Figure 2.1. I E E E reference model comparing with the ISO reference model  2.1.1  802.11 Topology The basic building block o f a 802.11 wireless L A N is a basic service set (BSS), which  consists o f stations that execute the same M A C protocol and compete for access to the same shared wireless medium.  A n independent B S S (IBSS) is a standalone B S S that has no  backbone infrastructure and consists o f at least two wireless stations as shown in Figure 2.2. This type o f network configuration is often referred to as an ad-hoc network. A B S S may be connected to a backbone distribution system through an access point ( A P ) to form an  8  extended service set (ESS) as illustrated in Figure 2.3. The term infrastructure network is used informally to refer to the E S S . The B S S s are like cells in a cellular network and the A P s are analogous to base stations in the cellular network [13]. This type o f configuration satisfies the needs of large coverage networks o f arbitrary size and complexity.  Figure 2.2. Independent basic service set  Figure 2.3. Extended service set  9  2.1.2  802.11 Protocol Architecture Figure 2.4 depicts the protocol architecture o f the 802.11 standard.  The channel  access for the wireless stations in a B S S is under the control o f the M A C layer with two sublayers, namely Distributed Coordination Function ( D C F ) and Point Coordination Function (PCF). D C F is a contention-based M A C protocol providing asynchronous data transfer on a best-effort basis and is a mandatory M A C protocol supported by all stations. control in ad-hoc networks uses D C F only.  The access  Infrastructure networks can operate using just  D C F or a coexistence o f D C F and P C F . Details o f the D C F operation w i l l be described in section 2.2.  802.2 Logical link control (LLC) I Contention-free service  t  Point coordination function (PCF)  MAC layer  Physical layer  Contention service  *  Distributed coordination function (DCF) 2.4-GHz frequencyhopping spread spreatrum  1  Data rates of 1 and 2 Mbps  5-GHz orthogonal frequencydivision multiplexing Data rates of 6, 9,12, 18, 24,36, 48, and 54 Mbps  2.4-GHz directsequence spread sprectrum Data rates of 5.5 and 11 Mbps  2.4-GHz orthogonal frequencydivision multiplexing Data rates up to 54 Mbps  802.11  802.11a  802.11b  802.1 lg  2.4-GHz directsequence spread sprectrum  Infrared  Figure 2.4. 802.11 protocol layers  10  P C F is a centralized M A C protocol providing contention-free multiple access service for the transmission of real-time traffic by polling stations i n turn. Studies have shown that it is difficult for P C F to achieve high efficiency and to satisfy the time requirement for real-time traffic, [14], [15] and the cooperation between D C F and P C F leads to poor performance [16]. P C F is not supported by commercially available wireless cards and A P s .  The P H Y specifies the actual transmission details and has been the focus of much work in the past few years. The 802.11 standard specifies several P H Y s . The initial standard approved in 1997 included frequency hopping spread spectrum (FHSS) and direct-sequence spread spectrum (DSSS) delivering data rates of 1 and 2 Mbps in the 2.4-GHz I S M band. The standard also defined an infrared option operating at 1 and 2 M b p s via passive ceiling reflection; one reason that the infrared option has never gained wide market support might be because infrared transmissions cannot penetrate walls.  Most of the early 802.11 wireless  L A N s are F H S S due to the relative implementation simplicity.  In 1999, the I E E E issued the second and third P H Y s , I E E E 802.11a and 802.11b, at roughly the same time. 802.1 l a operates i n the 5 - G H z U N I I band at data rates up to 54 M b p s using orthogonal frequency division multiplexing ( O F D M ) ; 802.11b operates in the 2.4-GHz I S M band with 5.5 and 11 Mbps data rates using D S S S . The I E E E has recently approved the final specification for 802.1 l g which adopts 802.1 l a ' s O F D M but runs 54 M b p s in the 2.4G H z I S M band and is backwards compatible with 802.11b. deployed today comply with the 802.1 l b standard.  11  Most 802.11 wireless L A N s  2.2  802.11 D C F Protocol The fAindamental access method o f the 802.11 M A C protocol is D C F , which supports  asynchronous data transfer on a best-effort basis and is the only possible M A C protocol in 802.11 ad-hoc networks. The 802.11 D C F M A C protocol operation is depicted in Figure 2.5.  Immediate access when medium is free >= DIFS • Contention Window -  •DIFS-  -DIFS-  n  — PIFS Busy Medium  —i—i—i—i—i—  -SIFS+  Backoff-Window  _i i i i i  Next Frame  h Slot time Select slot and decrement backoff as long  Defer Access  as medium is idle  Figure 2.5. 802.11 D C F M A C protocol operation  For a station to transmit a M A C protocol data unit ( M P D U ) , it must sense the medium to determine i f another station is transmitting and must ensure that the medium is idle for the specified D C F Interframe Space (DIFS) duration. A station may transmit a pending M P D U when it determines that the medium is idle for a time interval greater than or equal to the DIFS period. If the medium is sensed busy, the station continues to sense the channel for an additional random time after detecting the channel as being idle for the D I F S duration. The additional random time period is selected from C W . The size o f C W , bounded by the maximum value CW , max  is doubled after each unsuccessful transmission to reduce the  12  collision probability. value CW „. mi  Following each successful transmission, C W is reset to the minimum  This is the well-known binary exponential backoff ( B E B ) algorithm. The  backoff time, backoff time, can be expressed as follows [17]:  backoff _time = randInt(0,mm(CW  x2  reny  Tnin  ,CW ))xslot max  _time  (1)  In (1), randlnt(a, b) generates a random integer i n the range from a to b uniformly, min(c, d) gives the smaller value o f c and d, retry is the number o f retransmission attempts, and slot time is a time duration specified by the physical layer parameters.  During backoff, the station decreases its backoff counter by one i f the medium is idle for a slot time period and freezes the backoff counter when the medium is busy. When the backoff counter reaches zero, the station w i l l transmit its M P D U immediately.  When a  destination station receives the M P D U successfully, it returns an Acknowledgement ( A C K ) frame to the source station after a Short Interframe Space (SIFS) duration.  DCF  offers an optional means o f transmitting data frames that requires the  transmission o f Request T o Send ( R T S ) and Clear T o Send ( C T S ) frames prior to the transmission o f the actual data frames. The R T S / C T S transmission scheme can alleviate the hidden terminal problem and can reduce the transmission time wasted as a result of a collision due to the longer frame size o f the actual M P D U .  The R T S and C T S frames include  information regarding the transmission time o f the next data frame and the corresponding A C K frame.  The Network Allocation Vector ( N A V ) maintained by each station is an  13  indicator o f time periods when other stations close to the transmitting station and hidden stations close to the receiving station w i l l not commence any transmissions. D C F with the R T S / C T S transmission scheme and the N A V settings o f other stations are shown in Figure 2.6. SIFS Source  H—H DATA  RTS  SIFS  SIFS  SIFS  <—W Destination  DATA  CTS  SIFS ^ >\ Other  I  i i i i i i  NAV (RTS)  Contention Window  i i i i i i  NAV (CTS)  Backoff after defer  Defer Access  Figure 2.6. R T S / C T S / d a t a / A C K and N A V settings  2.3 New 802.1 le EDCF Protocol The new 802.1 l e E D C F medium access scheme is governed by a distributed mechanism very similar to 802.11 D C F . Service differentiation is achieved through the introduction o f Traffic Categories (TCs).  Each T C has a different transmission queue and  each transmission queue has a different interframe space (Arbitrary Interframe Space A I F S [ T C ] ) , a different set o f contention window limits ( C W „ [ T C ] and C W mi  different persistent factor (PF[TC]).  14  m a x  [ T C ] ) , and a  Figure 2.7 illustrates the service differentiation accomplished by using different A I F S values. Each T C within a station starts a backoff independently after detecting the channel as idle for an A I F S [ T C ] duration.  In the E D C F retransmission scheme, the size o f the new  C W [ T C ] after an unsuccessful transmission is determined by expanding/reducing the size o f the previous C W [ T C ] by a factor o f P F [ T C ] , whereas i n legacy 802.11 D C F , C W is always double after every unsuccessful transmission, i.e. PF=2. A s i n legacy D C F , the C W [ T C ] never exceeds its maximum bound C W  m a x  [ T C ] . A random backoff counter is chosen from the  interval [0, C W [ T C ] ] i n the case o f AIFS[TC]>DIFS and from [1, CW[TC]+1] i n the case o f AIFS[TC]<DIFS [18]. When the backoff counter o f a T C reaches zero, the station transmits a pending M S D U from the corresponding transmission queue. A short A I F S , a small size o f C W limits, and a l o w P F value are associated with high priority packets, enabling them to start contenting the channel earlier and to complete the backoff sooner, thus offering a high probability o f winning the contention race.  <4  AIFS[j] AIFS[i]  <  •)  DIFS/AIFS  Contention Window  PIFS DIFS/AIFS  N  H Busy Medium  H  SIFS  H  T71 _z  Backoff Window l  Next frame  l—  "Slot time  <  '  Defer Access  •  Select slot and decrement backoff as long as medium is idle  Figure 2.7. Service differentiation by different A I F S values  15  Within a station, up to eight transmission queues are realized as virtual stations. See Figure 2.8. Should the backoff counters o f two or more parallel T C s i n a single station count down to zero at the same time, a scheduler, which resides i n the station, resolves the virtual collision by allowing the highest priority T C among the virtually collided T C s to transmit its M P D U [19].  The other virtually collided T C s execute the retransmission mechanism  independently, as i f a collision had occurred.  802.11 one backoff instance  1r Backoff DIFS CW PF =2  802.1 le: up to 8 independent backoff instances PO  !  j| 1  | ]  • 1r Backoff AIFS[0] CW[0] PF[0]  r  PI  r Backoff AIFS[1] CW[1] PF[1]  r  P2  P3  r Backoff AIFS[2] CW[2] PF[2]  r Backoff AIFS[3] CW[3] PF[3]  P4  1r Backoff AIFS[4] CW[4] PF[4]  P6  P5  r Backoff AIFS[5] CW[5] PF[5]  r  r  Backoff AIFS[6] CW[6] PF[6]  Backoff AIFS[7] CW[7] PF[7]  1f  1f  1r 1r 1T r Scheduler (resolves virtual collision)  transmission attempt  P7  transmission attempt  Figure 2.8. Transmission architecture o f E D C F vs. D C F  Since 802.1 l e is a draft standard presently under review, many issues are still unsolved and are expected to change [8]. W e assume that E D C F described here w i l l not undergo any major modifications. The P F parameter mentioned i n this section was proposed  16  in.a previous version of the drafts but has been removed from the current draft [20].  In the  current version of E D C F , after each unsuccessful transmission, C W [ T C ] is doubled while remaining less than C W  m a x  [ T C ] , i.e. P F [ T C ] = 2.0 which is equivalent to the conventional  B E B algorithm.  17  Chapter 3 Age-Dependent Backoff Algorithm for 802.1 le EDCF The previous chapter has reviewed the 802.11 standard and has introduced the new 802.1 l e E D C F protocol, proposed by the 802.11 Task Group E , to provide service differentiation at the M A C layer o f 802.11 wireless L A N s . The retransmission mechanism o f the E D C F protocol causes large delay and jitter for real-time traffic.  W e now propose a  retransmission algorithm called Age-Dependent Backoff ( A D B ) to alleviate packet delay, jitter and drop rate for real-time traffic. The proposed A D B algorithm for the new 802.1 l e E D C F is presented i n this chapter. The backoff behavior o f 802.1 l e E D C F is described i n Section 3.1. Details o f A D B are provided i n Section 3.2. Finally, Section 3.3 discusses the complexity o f implementing the A D B algorithm.  3.1 Backoff Behavior of 802.1 le E D C F In legacy 802.11 D C F , the B E B algorithm doubles C W with every transmission retry to reduce the collision probability i n the next retransmission by providing greater transmission spacing among stations with pending M P D U s . C W becomes extremely large after successive retransmissions, which cause long delays and large jitter. T o reduce delay and jitter, a smaller C W should be employed i n the next retransmission thereby providing a better chance for retransmitted packets to access the medium. The wait for new arrivals would increase [21].  In a previous version o f the 802.1 l e draft, E D C F utilizes a multiplier, P F , to govern the adjustment o f C W after an unsuccessful transmission. P F should be less than 1 for timesensitive applications.  However, collisions are a result o f congestion and a wider C W is  18  desirable to alleviate congestion.  Reducing the C W size for every retransmission causes  heavy congestion leading to more collisions. Values o f P F between one and two would be preferable. Use of different PFs for different T C s contributes to service differentiation.  The current version of E D C F employs the B E B algorithm to resolve collisions. L i k e D C F , B E B causes large delay and jitter values that are problematic for real-time traffic with time-bounded requirements. The situation worsens when channel conditions are bad or when the traffic load becomes heavy.  3.2  Age-Dependent Backoff Algorithm  To alleviate the problem of the retransmission mechanism employed in 802.1 l e E D C F protocol, we now present the proposed Age-Dependent Backoff ( A D B ) algorithm for high priority real-time traffic. The idea of A D B is to dynamically adjust P F based on two factors, namely the age of a real-time packet i n the transmission queue and the lifetime of the realtime packet.  The relationship between the new C W , newCW[TC], and the old C W ,  oldCW[TC], after a collision is shown in (2).  newCW[TC] = ((oldCW[TC] +1) x PF[TC]) - 1  (2)  where PF[TC] is given in (3).  PF[TC] =  2 LT[TC]  19  Age+ 2  (3)  Age is the packet's age in the transmission queue and LT[TC] is the lifetime of the packet. Figure 3.1 shows the value of PF[TC] as the age of a real-time frame is varied. The newCW[TC] never exceeds the parameter C W  m a x  [ T C ] but can be less than C W j [ T C ] to m  n  provide differentiation between retransmitted packets and new arrivals.  Real-time packets are obsolete i f they are not received by recipients within their lifetime. Packets with queuing delay longer than the lifetime w i l l eventually be discarded by their applications and should not contend for the medium. Therefore, packets with Age > LT are discarded before attempting transmission to save bandwidth and to prevent causing additional delay to other packets.  PF[TC]  P-Age  LT[TC]/2  LT[TC]  Figure 3.1. PF[TC] vs. Age  It can be seen in (2) and (3) that the new C W is expanded by the factor 1 < P F < 2 in the first half of a packet's lifetime and compressed by the factor 0 < P F < 1 in the second half. During the first half of the lifetime, A D B allows the backoff to increase gradually while  20  avoiding high collision probability, but at the same time precluding a huge increase o f delay and jitter. During the second half o f the packet's lifetime, the backoff decreases slowly from the expanded C W to raise transmission probability, thereby preventing packets from being dropped.  It should be emphasized here that the A D B algorithm is used together with the B E B algorithm i n 802.11 wireless L A N s .  The A D B algorithm is for the high priority real-time  traffic with time-bounded requirements, while the B E B algorithm is for the best-effort traffic with tolerance o f large delay and jitter. A D B provides higher access priority than B E B , since A D B expands C W by a factor between 1 and 2 or reduces the C W by a factor between 0 and 1, whereas B E B always expands C W by a factor o f 2.  A D B contributes to service  differentiation by allowing a real-time frame with an age close to its lifetime to have higher priority for channel access than does a new arrival real-time frame.  3.3 Implementation Complexity The A D B retransmission mechanism is easy to implement.  It requires minor  modifications i n the computation o f C W thereby minimizing the migration effort from the new 802.1 l e E D C F mode. Our A D B retransmission strategy provides backward compatibility to the 802.11 D C F protocol.  3.3.1  Implementation of the ADB Algorithm Implementing the A D B algorithm is relatively simple. T o keep track o f the ages o f  21  the frames i n each transmission queue, every pending frame i n the queue requires a time stamp to record its arrival time from the upper layer. A temporary timestamp field can be appended to the 802.11 M A C frame to hold the arrival time. The timestamp field should be removed just before the actual transmission to keep the protocol backwards compatible with the 802.11 standard. The variable Age i n (2) is calculated by the current time o f the system minus the time stamp o f the frame at the head o f the queue.  If Age > LTVTC], the  corresponding frame w i l l be dropped and the next frame i n the queue w i l l be evaluated. The process continues until a frame with Age < LT\TC] is encountered or until there are no frames in the queue. If the queue holds no frames, the corresponding queue w i l l not contend for the access o f the channel.  The newCW[TC] is determined using (2) and (3). The smaller value o f newCW[TC] and C W  m a x  [ T C ] is chosen to be the C W size. After the new C W size has been determined,  the value o f oldCW[TC] is updated and w i l l be used for the calculation o f the next C W size.  The flow chart shown i n Figure 3.2 illustrates the implementation o f the A D B algorithm.  22  START A collision occurred in previous transmission and the channel is idle for a period of AIFS[TC]. Need to adjust the new C W.  Calculate Age Age = current time - timestamp  YES  YES  Drop the frame  NO  NO  END  Calculate PF\TC\ PP\TC] = (2/LT[TC]) x Age  The corresponding queue will not contend for the access of the channel  r Calculate newCW\TC\ newCW[TC\ = ((oldCW[TC\+\) x PF[TC\) - 1 r  if newcw\rc\ > c w ^ i r q ?  YES  Check to see if the new CW is larger^ than the maximum size^  r  Set newCW\TC\ to the maximum size newCW[TC\ = CW [TC] max  NO  Update oldCW\TC\ oidcwyrc] = newCW[TC] 1r  END Starts backoff  Figure 3.2. Flow chart for the implementation o f the A D B retransmission algorithm  23  3.3.2  Discussion of the Implementation Complexity The implementation o f the A D B algorithm described in the previous section is  practical with current software and hardware technologies and requires very few additional resources. Most o f the modern microprocessors are equipped with real-time clocks offering timer functions; these can be used to complete the timestamp field and to provide system current time. The calculation o f Age requires one subtraction. One comparison is needed to check i f Age > LT\TC] and one register is required to buffer the parameter LT\TC\ for each transmission queue.  The calculation o f PF[TC]  in (3) requires one division, one  multiplication, and one addition and the calculation o f newCW[TC] i n (2) introduces one addition, one subtraction, and one multiplication.  The additional resources mentioned above are necessary to implement the A D B algorithm. The addition, subtraction, multiplication, division, and comparison operations can be done using digital hardware circuits or software instruction codes. The A B D algorithm can be implemented with minor modification o f the existing hardware or the embedded software of the wireless cards.  24  Chapter 4 Design of Simulation Models The discussion in the last chapter has presented the problem o f the retransmission mechanism employed i n the new 802.1 l e E D C F protocol and has introduced our proposed A D B algorithm for real-time traffic i n 802.1 l e E D C F . T o evaluate the performance o f A D B against B E B , several verification methods can be used. One method is to implement the A D B algorithm in wireless cards and to then measure the delay, jitter and drop rate o f real-time packets from an actual network. This is definitely a high-cost and time-consuming approach. A n alternative efficient method is to use computer simulation. W e follow this widely used method, based on O P N E T simulations, to evaluate the performance o f A D B against B E B .  In this chapter, we review the O P N E T simulation tools i n section 4.1. Section 4.2 describes the simulation models for the 802.11 standard. Section 4.3 discusses the simulation models for the upper layer protocols. Section 4.4 presents the voice, the video, and the data traffic source models used i n our work.  4.1  Overview of O P N E T Simulation Tools Optimized Network Engineering Tools ( O P N E T ) , licensed by M i l 3 , Inc., is a piece o f  engineering software capable o f simulating large communication networks with detailed protocol modeling.  It is widely accepted by researchers to model and simulate complex  network scenarios, communication protocols, and traffic models. While O P N E T has many pre-defined models, it allows users to build new models by using finite state machines  25  together with the Proto-C language, a combination of general C/C++ facilities and an extensive library of built-in and high-level subroutines known as Kernel Procedures.  OPNET allows users to define network topologies, nodes, and links that describe a network. A simulation can be executed and the results are then analyzed for the network. OPNET has three main types of tools, namely the Model Development tool, the Simulation Execution tool and the Result Analysis tool. These three types of tools are used together to model, simulate and analyze the network.  OPNET defines a network model using a hierarchical structure shown in Figure 4.1. The highest level is the network domain, which is constructed from the node domain and the link domain. The node domain is specified in terms of the process domain.  Network Domain  Node Domain Link Domain Process Domain  Figure 4.1. Hierarchical structure for OPNET models  26  The Network Domain defines a network topology described in terms of subnetworks, nodes, links, and geographical context.  It consists of the nodes and the links which can be  deployed within the geographical context. O P N E T provides fixed nodes as well as point-topoint and bus links. In addition, it offers mobile and satellite nodes, and wireless links.  The Node Domain provides for the modeling of communication devices that can be deployed and interconnected at the Network Domain. The device models are called nodes and may correspond to various types of computing and communicating equipment in the real world such as routers, switches, bridges, workstations, servers, etc.  The Process Domain allows users to create processes using finite state machines and the Proto-C language.  These processes are used to express the behavior of protocols,  algorithms, and applications.  O P N E T comes with built-in model libraries for some common protocols and enables users to focus on the modeling of user-defined protocols.  The wireless L A N simulation  model is shipped as part of the standard O P N E T model and is based on the I E E E 802.11 standard. Specific parts of the wireless L A N model have been simplified or omitted in view of the fact that it is intended primarily for the performance estimation of the 802.11 D C F protocol. W e implement the new 802.1 l e E D C F protocol and the proposed A D B algorithm by modifying the existing wireless L A N model. The detail description of the modification w i l l be presented i n the next section.  27  4.2 802.11 Model Design Our 802.11 L A N model is implemented in O P N E T by modifying the existing wireless LAN  model to support the new 802.1 l e  EDCF  protocol  and our proposed A D B  retransmission algorithm. In this section, we describe the 802.11 model from the bottom to the top, i.e. from the physical layer to the M A C layer.  4.2.1 802.11 Physical Layer Model The primary function o f the 802.11 physical layer is to transmit a sequence o f bits over the wireless medium. The I E E E 802.11 High Rate Direct Sequence Spread Sprectrum ( H R - D S S S ) physical layer, commonly referred to as 802.1 l b , is a rate extension to the 802.11 D S S S standard.  It operates i n the 2 . 4 G H z I S M band and includes complementary code  keying ( C C K ) to achieve additional data rates o f 5.5 and 11 Mbps.  802.11b is the most  common wireless L A N implementation today and is interoperable with 802.11  DSSS  implementations. W e design our 802.11 physical layer model to be similar to the 802.1 l b .  The existing wireless L A N model provides three choices for the physical layer configuration i n the I E E E 802.11 standard: Frequency Hopping, Direct Sequence, and Infra Red. The wireless L A N model supports data transfer at 1, 2, 5.5 and 11 Mbps. A l l control packets are transmitted at a data rate o f 1 Mbps as specified by the standard. These data rates are modeled as the speed of the transmitter and receiver connected to the 802.11 M A C layer. Although the wireless L A N model does not simulate the actual physical layer o f the I E E E  28  802.11 specification, the physical layer configuration are needed by the M A C protocol to determine the parameters such as SIFS, D I F S and C W limits.  In our study, the modulation technique is relatively unimportant to the service differentiation in 802.11 wireless L A N s . However, the effect of wireless channel noise must be considered due to its impact on bit and packet errors which cause retransmissions. Because of transmission impairments, such as noise and collisions, bit errors can disrupt the sequencing of frames. A station may send a data frame and never receive an A C K . Stations initiating the exchange of frames have the responsibility of error recovery.  This recovery  involves the retransmission of frames after SIFS s e c , i f no response is heard from the destination station. Therefore, a frame corrupted by noise or which has collided with another frame causes execution of the retransmission protocol.  The O P N E T Wireless Module allows users to specify the quality of a wireless link based on a number of factors such as the distance between the transmitter and the receiver, the power of the transmitting signal, the transmitter  and the receiver antenna gains, the  background and the interference noises, etc. O P N E T ' s simulation kernel relies on a 14-stage computational pipeline which uses these factors to emulate the characteristics of the wireless communication channel.  The bit error rate can be measured statistically during the  simulation.  The existing O P N E T wireless L A N model uses the default wireless model settings with some minor modification to conform to the 802.11 standard. W e specify the noise figure  29  parameter, the transmitting power, and the distance between the transmitting and receiving stations to achieve a bit error rate of 10" . W e set the error threshold to be zero. Hence, any 9  errors in a packet are discarded by the M A C layer.  In order to model the 802.1 l b physical layer, we set the channel data rate at 11 M b p s with direct sequence spread spectrum physical characteristic.  W i t h this direct sequence  physical characteristic, we specified SIFS = 10 u.s, D I F S = 50 u,s, C W j = 31, and C W m  n  m a x  =  1023.  4.2.2  802.11 M A C Layer Model Based on the existing wireless L A N model in O P N E T , we modify the 802.11 D C F  protocol at the M A C layer to develop the new 802.1 l e E D C F protocol with the proposed A D B algorithm. MAC  The existing wireless L A N model is intended primarily for DCF-based  performance  estimation.  Many  important features  of the  802.11  M A C are  implemented. The following is a summary of the important features included in the wireless L A N model [22].  •  Access  Mechanism:  Carrier  sense multiple  access  and collision avoidance  ( C S M A / C A ) D C F access scheme as defined i n the standard. •  Frame exchange sequence: Data and A C K frame exchange to ensure the reliability of data transfer. Optional R T S / C T S frame exchange is available for media reservation.  30  •  Backoff and Deference: The B E B algorithm and the interframe spacing: SIFS, D I F S , and EIFS for D C F implementation. The values for the interframe spacing are selected based on the selection o f physical characteristics.  •  Channel Data Rate: Date rates of 1, 2, 5.5 and 11 M b p s  •  Recovery mechanism: Short and Long retry counters as defined in the standard.  •  Fragmentation and Reassembly: Optional data frame fragmentation based on the size of the data packet received from the higher layer. The fragments are reassembled at the destination station.  •  Duplicate Packet Detection: A n y duplicated packets are discarded by the M A C layer.  •  Access Point Functionality:  Only a wireless L A N router can be configured as an  access point in an infrastructure B S S network.  In order to form an E S S such that  stations within various B S S s can communicate with each other, an IP network needs to be configured. •  Buffer Size: Data arrived from a higher layer to the M A C layer is stored in a buffer. The buffer size is limited to the maximum value set by the user. Higher layer packets are dropped, once the maximum buffer size is reached.  Some additional features are necessarily added to make that the existing wireless L A N model supports E D C F and A D B . The wireless L A N model has only one transmission queue buffering frames waiting for transmission.  A l l packets coming from the upper layer are  forwarded directly to that transmission queue.  The packets placed in the queue w i l l be  transmitted only i f the backoff counter reaches zero and the wireless channel has been idle for  31  DIFS sec. The model has only one backoff counter and one interframe spacing period, D I F S , for all data frame transmissions.  Multiple transmission queues, backoff counters, C W limits and interframe spacing periods are necessary for implementing the E D C F protocol. Packets arriving from the upper layer are forwarded to the corresponding queues based on the traffic types. T o simulate a wireless L A N carrying voice, video, and data traffic, we implement three transmission queues for three different traffic types.  In order to simplify the implementation, each station is  allowed to generate and receive only one type of bi-directional traffic and therefore, using destination M A C addresses is enough to determine the traffic types.  A scheduler, which  resides in an A P , is implemented to resolve the virtual collision when two or more T C s in the access point ( A P ) count down to zero at the same time.  To implement the A D B algorithm, each pending frame is given an arrival timestamp. A l l packets from the upper layer are sent to the corresponding transmission queues with their timestamps which are used to calculate Age,  as explained in section 3.3.1.  The  implementation of A D B in the simulation model is as presented in section 3.3.1.  The finite state machine for modeling the 802.11 M A C layer with E D C F and A D B is shown in Figure 4.2. A transition between the B K O F F N E E D state and the I D L E state is added to the finite state machine. This transition is used when the ages of all the packets in the transmission queues are greater than their lifetimes.  The expired real-time packets are  then dropped from the queues since packets with ages greater than their lifetimes w i l l  32  eventually be discarded by the applications. The station should not contend for the access o f the channel and the finite state machine should go back to the I D L E state.  f ^( d«fault) ^^fc< | H I TT 1|  »|B33_IHIT|  »f IDLE \[*  <FRM_EHD_TO_D ILE> | ^_(IpLE_AFTER_CFP ) / CARCEL_DEF_EVEHT;  ^  P  (READ¥JTO_TRAHStoTl 6A f&BHJVM_3 I _D I LE)  Tr-  ^^T^P^^  F (RAME_TW I EOUT || FRAME_RCVD)  w  ^v-"""  h V (AT I_FOR_FRAME)  ^  (default)  y ^ <" "F (RM_EHD_T 0_DEFER)  (BACK_T 0_DEFER)/wa lr\_s cite due l ^de f e reno e ( ) ! (default)  (HIS SED_DEAELH I E)\\  (DEFEBEflCE_0FF)  T (PJU1 SMS I SI OH_C OBPLETE ) ^mbp_/EED] &&  (PERF 0RM_BACKOFF  (BACK_TO_D I LE)  (TRAH S M I T _ F R A M E &£.  !MISSED_DEADLIHE;  'fas SED_DEADLXHE )  \FP (jlERFORM TRANSMIT)/wa l np j erf oim._transifcit_ch«ck () HTRBHSMT |BACKOP I|  ^.'^ (de£*ult)/wl»n_ijat«rrupts_in£o (); (default) R ( EADY_TO_TRH fl SMT I MEDV IM_IS_D I LE £•& 0 fP_»P_»G duim_C ontro 1 -= OP C_B 0 OLH I T _DI SABLED ) Figure 4.2. Finite state machine for 802.11 M A C layer model with E D C F and A D B  The INIT and the B S S I N I T states are used to initialize all the variables, statistics, memory allocations and other entities.  When a packet from the upper layer arrives and the  channel is idle, the state machine w i l l go directly to the T R A N S M I T state to transmit the packet. If a packet arrives but the channel is busy, the state machine w i l l move to the D E F E R state to defer for a period o f D I F S / A I F S after the channel becomes idle.  After  the deference  described above, the state  machine  will  jump  to the  B K O F F N E E D state. A C W size is then calculated using the B E B or the A D B algorithm.  33  Once the C W size is determined, the state machine w i l l go to the B A C K O F F state to udergo the backoff operation. When a backoff counter reaches zero, the state machine w i l l move to the T R A N S M I T state to transmit the packet. After finishing the transmission, the transmitting station w i l l wait for an A C K frame from the receiving station to confirm the success of the frame reception.  The state machine w i l l jump to the F R M E N D state and then to the  W A I T F O R R state to wait for the A C K frame. If the A C K frame arrives successfully, it w i l l go either to the I D L E state through the F R M E N D state i f there are no pending frames i n the transmission queues or to the D E F E R state through the F R M E N D state to start the next transmission i f the transmission queues are not empty.  Table 4.1 shows the simulation parameters for the M A C layer protocol.  In our  simulation scenarios, detailed in Chapter 5, we assume that the simulation network consists of one B S S with no hidden stations, i.e. stations can hear each other in the B S S .  The R T S / C T S  optional is disabled since it has been indicated that the R T S / C T S mechanism produces very limited advantages with no hidden stations [23].  Moreover, most of the default settings o f  commercially available wireless cards are without the R T S / C T S mechanism.  The short retry limit gives the maximum number of transmission attempts for frames whose size are less than or equal to R T S threshold, while the long retry limit specifies the maximum number o f transmission attempts for frames whose size exceeds the R T S threshold. When the R T S / C T S is disabled, the short retry limit is used for all transmissions. Frames that could not be transmitted successfully after reaching the short retry limit are discarded by the M A C layer.  34  Values  Parameters Slot time  20 p.s  SIFS  10 us 10u.s + 2 x 2 0 u.s = 50 \is  DIFS AIFS[1] (voice)  10|j.s + 2 x 2 0 p.s = 50 p,s  AIFS[2] (video)  10 u.s + 3 x 20 u,s = 70 [is  AIFS[3] (data) rCWmin, C W m a x l ( D C F ) [ C W m i n [ l ] , CWmaxI/l]] (voice) [CWmin[2], C W m a x [ 2 H (video) rCWmin[3], CWmax[311 (data) Short Retry Limit Long Retry Limit  10p.s + 5 x 2 0 p,s=110p,s T31, 10231 r7,3ii [15,63] [15,255] 255 255  Table 4.1. Parameters for the M A C layer  The A D B algorithm is used for the retransmission mechanism for real-time traffic with time-bounded requirements.  The lifetime of a real-time packet must be specified to  allow A D B to calculate the new C W size after every unsuccessful transmission. We now discuss the lifetimes of voice and video real-time packets.  The main performance parameters for voice traffic are packet delay, jitter, and loss. The end-to-end delay for IP telephony ranges from 300 to 1000 ms. User tolerance o f delay varies significantly.  Demanding users require delay of 200 ms or less, while more patient  users w i l l accept a delay of 300 to 800 ms [24].  Since the wireless L A N represents only a  single hop of an end-to-end connection, we consider 25 ms as the maximum acceptable value for the voice packet transfer delay over the wireless L A N and let LT for voice packets be 25 ms.  35  W e assume that 75 ms is the maximum allowable video packet delay at a single wireless hop; thus LT = 75 ms for video packets.  4.3  Simulation Models for Upper Layer Protocols The existing O P N E T wireless model includes node models which correspond to  various real world wireless L A N equipment types such as wireless stations and wireless routers. B y replacing the original M A C layer with our modified M A C layer presented i n the previous section, we developed wireless station and router node models supporting the new 802.11 E D C F protocol and the proposed A D B algorithm. The upper layer protocols such as the Internet Protocol (IP), the Transport Layer Protocol (TCP), the User Datagram Protocol ( U D P ) and the application layers are unmodified.  In this section, we introduce the basic  functions and parameters chosen for these upper layer protocols from bottom to top, i.e. from the network layer to the application layer.  4.3.1  Network Layer Model: Internet Protocol IP is a connectionless network level protocol interconnecting multiple  possibly o f different types.  networks,  In the Internet Protocol Suite, IP provides best-effort delivery  services to T C P and U D P and is situated beneath the transport layer o f the OSI seven-layer model [13]. In turn, IP relies on the services o f the data link layer, such as the 802.3 Ethernet and the 802.11 wireless L A N , to relay packets to other IP modules.  The packets that are created and forwarded by IP modules are called datagrams. Because IP connects different types o f networks that may support different maximum transfer  36  units ( M T U s ) , fragmentation is performed by the IP modules whenever they attempt to forward a datagram over an interface whose datagram length exceeds its M T U . Reassembly functions are only performed on datagrams when they reach their final destination.  The M T U of 802.11 wireless L A N s is 2304 bytes.  W e configure the IP modules to  perform fragmentation when forwarding a datagram with size larger than 2304 bytes over an 802.11 interface. Since the size of the data packets from the IP layer cannot be larger than the 802.1 l ' s M T U , the fragmentation and reassembly option at the 802.11 M A C layer can be disabled.  4.3.2  Transport Layer Models: T C P and UDP Two transport layer protocols, T C P and U D P , are build on the best-effort service,  provided by IP, to support a wide range of applications.  In this section we discuss the  properties of T C P and U D P and the parameters chosen in our simulations.  U D P is a very simple but unreliable, connectionless transport layer protocol that provides only two additional services beyond IP: demultiplexing and error checking on data. IP delivers packets to a host but does not know how to deliver them to the specific application in the host. U D P adds a mechanism that distinguishes among multiple applications within the host. IP only checks the integrity of its header. U D P can optionally check the integrity of the entire U D P datagram. If a datagram is found to be corrupted, it w i l l be simply discarded and the source U D P entity w i l l not be notified.  37  T C P is a transport layer protocol, providing a connection-oriented, reliable, i n sequence, byte-stream service. T C P offers flow control that allows receivers to control the rate at which the sender transmits information so that the receiver buffers do not overflow. T C P also provides congestion control which ensures that the sender does not waste resources by sending more traffic than the network can successfully carry to the receiver. Similar to U D P , T C P supports multiple applications in a host and error checking on T C P segments.  T C P and U D P both have the ability to establish a host-to-host communication channel for delivering packets between applications running in two different stations.  The main  difference is that T C P provides reliability, and flow and congestion control services, while U D P trades off those services to improve performance.  Application protocols can choose  T C P or U D P at the transport layer. The File Transfer Protocol (FTP) uses T C P to ensure that an exact copy of a file is delivered to the recipient. Real-time applications such as voice and video conferencing use U D P because they can tolerate some errors or data loss.  O P N E T ' s T C P module is based on the T C P specified in R F C 793 and R F C 1122. The model incorporates some important  features such as end-to-end reliability  based on  acknowledgment, retransmissions triggered by exponentially backed-off timers, flow control based on the availability of remote buffering resources, reordering of data that arrives out of sequence, connection establishment and closing through three-way handshake protocols, and "Slow-start" congestion avoidance and control [25].  The model allows users to control the  T C P behavior by selecting the T C P parameters. W e choose the default setting o f the T C P parameters. Some of the important default parameter settings are shown in Table 4.2.  38  M a x i m u m Segment Size Receive Buffer M a x i m u m A C K Delay Slow-Start Initial Count Fast Retransmit Fast Recover Karn's Algorithm Initial R T O Minimum R T O Maximum R T O R T T Gain R T T Deviation Gain R T T Deviation Coefficient Timer Granularity Persistence Timeout  2272 bytes 8769 bytes 0.200 sec 1MSS Enabled Reno Enabled 3.0 1.0 64 0.125 0.25 4.0 0.5 sec 1.0 sec  Table 4.2. Default settings of the T C P parameters  The maximum segment size is set to 2272 bytes so that the underlying 802.11 network can carry traffic without any fragmentation. W i t h the fast retransmit algorithm enabled, after receiving three duplicate A C K s , the T C P sender infers that a segment is lost and re-transmits the missing segment without waiting for its retransmit timer to expire [26]. This allows T C P to detect and retransmit a lost segment more quickly than for time-out retransmission. T C P Reno employs Karn's algorithm [27], which requires round-trip estimates to be updated only for the A C K s of those segments that have been transmitted only once. T C P uses frequent measurements of the round trip timer ( R T T ) to dynamically adjust the retransmission timeout (RTO).  When the R T O expires before an A C K is received, retransmission occurs starting  with the first byte of unacknowledged data.  The R T T gain and R T T deviation gain are  constants between 0 and 1 and control how fast the R T O adapts to changes in network traffic. The R T T gain and the R T T deviation gain are recommended to be 0.125 and 0.25 respectively  39  [28].  For further information regarding the default setting, please refer to the TCP Model  Description of OPNET user's manual [25].  In OPNET's UDP model, there are no parameters to specify the behavior of UDP since UDP itself is a very simple protocol. If the application layer sends a packet with size larger than the maximum payload size of a UDP datagram, 65535 bytes - 8 bytes = 65527 bytes, the packet will be dropped.  4.4 Traffic Source Models We use OPNET's standard network application model for the traffic source models. The standard network application model is a simple, general model of client and server or a peer-to-peer network application. Its behavior can be modified through parameter settings to make it emulate a wide variety of network applications.  It does not, however, model the  behavior of any particular application in detail.  The parameters of the standard network application model are selected by carefully emulating the environment in which 802.11 M A C would most probably be applied. With this in mind, we choose numerical parameters in such a way that the traffic sources reflect as closely as possible the network load conditions of an actual real-life situation.  The simulations consider three types of traffic sources: voice, video conferencing and FTP traffic. Each wireless station runs only one session and all sessions are bi-directional, i.e. each station is the source of an uplink flow and the sink of a downlink flow for the session it  40  runs.  V o i c e and video conferencing traffic have higher priority than F T P traffic.  Section  4.4.1 and section 4.4.2 describe the voice and video conferencing models, respectively. The best-effort F T P file transfer model is presented in section 4.4.3.  4.4.1  Voice Model V o i c e traffic is known to have a two-state O N / O F F behavior, where voice users are  either in talkspurt or silence as shown in Figure 4.3.  For efficient usage of wireless  bandwidth, silence suppression is employed. Hence, voice packets are generated in talkspurt (ON), while no packets are generated during silence (OFF). Both the duration of talkspurt and of silence follow the exponential duration with the mean duration equal to 1 and 1.35 seconds respectively [29]; thus a voice source is in talkspurt mode approximately 42.6% of the time. Each voice station runs only one bi-direction voice session over U D P / I P .  Figure 4.3. Two-state model of a voice user  The G.729 speech codec is selected to model good-quality voice calls, with 8-kbps coding rate, 10-byte packets generated i n every 10 ms during talkspurt, 10 ms processing delay, and 5 ms lookahead delay.  The processing delay is the delay required to run the  41  encoding algorithm on a single frame and the lookahead delay is the amount o f time needed to delay the speech for look ahead calculations for the encoder frame. The encoder buffers data for this extra time to enable prediction, while compressing data. The one-way latency o f the encoder is the sum o f the frame size, processing delay, and lookahead delay and therefore, the total codec delay for G.729 is 25 ms [30].  4.4.2  Video Conferencing Model The video conferencing application i n O P N E T ' s standard network application model  is used to generate streaming video frames across the wireless L A N . It allows users to control the properties o f streaming video through the application parameters. Table 4.3 shows the parameters selected for modeling a low-quality video conferencing application characterized by a relatively l o w bit rate o f 128 kbps for both the uplink and downlink  Incoming Stream File Size  Description Time between frames generated within a video conferencing session from the destination Time between frames generated within a video conferencing session from the source Average size o f an incoming video frame  Outgoing Stream File Size  Average size o f an outgoing video frame  Parameters Incoming Stream Interarrival Time Outgoing Stream Interarrival Time  Values 20 frames/sec  20 frames/sec  Exponentially distributed with a mean o f 800 bytes Exponentially distributed with a mean o f 800 bytes  Table 4.3. Parameters for video conferencing  42  UDP transport protocol is used for the video conferencing application. A video station runs one bi-directional video session over UDP/IP.  4.4.3 FTP File Transfer Model An FTP application enables file transfers between a client and a server.  The FTP  application in OPNET's standard network application model is used to generate best-effort data traffic across the network. FTP has two basic commands for transferring a file: Get and Put. The Get command triggers the file transfer from the server and the Put command sends a file from the client to the server. FTP uses the connection-oriented TCP transport protocol. A new transport connection is opened for each file transfer. Table 4.4 shows the parameters chosen in the FTP application.  Parameters Command Mix (get/total)  Inter-Request Time  File Size  Description Ratio of the Get (download) commands to the total number of commands (sum of Get and Put commands) Time between subsequent file requests  Size of a file being transferred  Values 50%  Exponentially distributed with a mean of 0.02048 second 1024 bytes  Table 4.4. Parameters for FTP file transfers  43  Using the above parameter values, we assume that the upload and the download streams are identical with 2 0 0 kbps of FTP traffic in each stream. The rate at which files are requested is independent of the response received; that is, the second request can initiate without the first response being received.  44  Chapter 5 Performance Analysis of the Proposed ADB Algorithm in an Ad-Hoc Network and a Hotspot The previous chapter has introduced the OPNET simulation models.  We will use  these models to evaluate the performance of the new 802.1 le E D C F protocol with our proposed A D B retransmission algorithm. The behavior of E D C F without A D B is presented for comparison.  This chapter presents simulation parameters and scenarios, results from the  simulations, and discussions of the results.  Two typical environments of 802.11 wireless LANs are considered, namely the ad-hoc network and the hotspot scenarios. In the ad-hoc network scenario, different services such as voice, video and data services may be active simultaneously in an ad-hoc network. The adhoc network is an independent BSS in a restricted space where stations can hear each other.  The hotspot scenario is conceived where voice, video, and data users are connected to wireline networks through an AP in a single BSS.  A hotspot is a specific type of  infrastructure mode installation in which a commercial entity provides wireless Internet access to the public. Any reference herein to "hotspot" installations would apply equally to other infrastructure mode installations.  Section 5.1 describes the ad-hoc scenario and section 5.2 presents the hotspot scenario. Both sections provide simulation results as well as discussions concerning the results. A l l results presented in this chapter are the average of results collected from simulating each of  45  the scenarios for 180 seconds for each of 5 different seed numbers. Section 5.3 provides a conclusion regarding the simulation results.  5.1 Ad-Hoc Network Scenario This section presents the ad-hoc network scenario.  The overview of the ad-hoc  network scenario is introduced in section 5.1.1. The simulation results are given in section 5.1.2. The discussions are provided in section 5.1.3.  5.1.1  Overview of the Ad-Hoc Network Scenario  One environment in which 802.11 wireless L A N s is likely to be applied is in an adhoc network.  Such situation is appropriate for, assemblies, outdoor activities, rescue  operations, or major disaster areas, where users need to deploy networks to start communication immediately without the benefit of a fixed network infrastructure.  Figure 5.1 shows the topology of the ad-hoc network. For simplicity, we assume that there are no hidden stations and that the distance between every pair of transmitting and receiving stations is the same. The bit error rate of each wireless link is therefore assumed to be constant and equal. In the ad-hoc network, voice, video conferencing and best-effort FTP file transfer services are active simultaneously.  We conceive that the ad-hoc network is an independent BSS with 10 voice, 4 video and n FTP client and server stations where n is variable. Since 802.1 le E D C F is designed to  46  be backwards compatible to 802.11 DCF, we assume that half of the FTP client and server stations are using the 802.11 D C F while the other half are using the new 802.1 le EDCF protocol with the BEB algorithm with PF value of 2.0. We simulate the ad-hoc network to compare, for real-time traffic, the performance of our A D B algorithm with that of the generalized B E B algorithm with PF = 1.5 and 2.0.  Best-effort FTP data traffic always  employs the conventional BEB algorithm where PF = 2.0.  Independent basic sevice set  n/4 FTP clients ^ (802.1 le EDCF)  n/4 FTP servers (802.lie EDCF) 200 kbps of FTP traffic per flow  ^  n/4 FTP clients (802.11 DCF)  802.11 ad-hoc network  Figure 5.1. Network topology of an ad-hoc scenario  5.1.2 Simulation Results from the Ad-Hoc Network Scenario  This section presents the results of the ad-hoc network scenario. Voice packet delay, jitter, and drop rate are shown in Figures 5.2, 5.3, and 5.4 respectively. We define the jitter as  47  the delay variance and the drop rate as the percentage of packets with delay longer than their lifetime. The lifetime of a voice packet is assumed to be 25 ms.  12  16  20  24  28  number of FTP stations  -ADB  • EDCF (PF=2.0)  • EDCF (PF=1.5)  Figure 5.2. Voice packet delay in an ad-hoc network  0.0007  12  16  20  24  number of FTP stations  -ADB  • EDCF (PF=2.0)  • EDCF (PF=1.5)  Figure 5.3. Voice packet jitter in an ad-hoc network  48  28  12  16  20  number of FTP stations -ADB  - E D C F (PF=2.0) — A — E D C F  (PF=1.5)  Figure 5.4. Voice packet drop rate in an ad-hoc network  Figures 5.5, 5.6, and 5.7 show delay, jitter and drop rate, respectively, for video packets. The lifetime of a video packet is assumed to be 75 ms and therefore, a video packet is dropped if its age exceeds 75 ms.  8  12  16  20  24  number of FTP stations -ADB  - E D C F (PF=2.0) — A — E D C F  (PF=1.5)  Figure 5.5. Video packet delay in an ad-hoc network  49  28  Figure 5.6. Video packet jitter in an ad-hoc network  The main performance characteristic of the best-effort F T P traffic is measured by throughput. The total F T P traffic throughput for the ad-hoc network is shown in Figure 5.8.  number of FTP stations -ADB  •E D C F  (PF=2.0)  •E D C F  (PF=1.5)  Figure 5.7. Video packet drop rate in an ad-hoc network  50  700000 g-  600000  5  500000  Q. o>  400000  1  300000  •is  200000  «  100000  o  0 4  8  12  16  20  24  28  number of FTP stations — • — A D B  - » - E D C F  (PF=2.0) - A - E D C F  (PF=1.5)  Figure 5.8. Total FTP data throughput in an ad-hoc network  5.1.3 Discussions of the Ad-Hoc Network Results Our simulation results demonstrate that EDCF with the A D B algorithm outperforms EDCF with the generalized B E B algorithm for PF = 1.5 and 2.0 in all aspects of network performance that we have tested.  The voice packet delay, jitter and drop rate are significantly lower when using A D B as illustrated in Figures 5.2, 5.3, and 5.4. Relative to BEB, A D B also demonstrates considerable improvements in the video packet delay, jitter, and drop rate as depicted in Figures 5.5, 5.6 and 5.7.  The relative improvements in voice and video traffic become more noticeable and pronounced as the number of FTP stations increases. A D B enhances the performance of the  51  voice and video traffic without sacrificing the throughput of the best-effort FTP traffic; as well, A D B prevents the FTP traffic from being starved as shown in Figure 5.8.  A D B dynamically adjusts the change of C W sizes based on the ages and lifetimes of voice and video packets to avoid long delay and to prevent excessive collisions under heavy traffic load. As a result, the delay, jitter and drop rate of both voice and video packets are improved considerably. These enhancements in voice and video packet performance do not adversely affect the FTP traffic throughput. Accordingly, we can conclude that A D B enables more efficient use of the wireless channel and improves the overall performance of the ad-hoc network.  5.2 Hotspot Scenario  The subject of this section is the hotspot scenario.  A n overview of this scenario  appears in section 5.2.1. Simulation results are given in section 5.2.2, and discussions are provided in section 5.2.3.  5.2.1  Overview of the Hotspot Scenario A hotspot is another environment where 802.11 wireless LANs would likely be  deployed. A hotspot is a geographic area covered by wireless networks, with Internet access being available to those devices with wireless network cards.  52  Most of the hotspots today are covered by 802.1 lb wireless LANs to provide wireless Internet access services for public use.  The network topology associated with the hotspot  scenario is depicted in Figure 5.9.  Figure 5.9. Network topology of a hotspot scenario  The hotspot scenario is developed to study the performance of the A D B algorithm in a hotspot environment. For simplicity, we assume that there is only one AP in the hotspot and that there are no hidden stations in a BSS. The distance between each wireless station and the AP is the same and constant, in which case the bit error rate of each wireless link is assumed to be identical.  Voice, video conferencing and best-effort FTP file transfer services are active simultaneously in the hotspot. Because users in the hotspot environment rarely make phone  53  or video conferencing calls, or rarely transfer files to their neighbors, all types of traffic generated within the BSS are delivered to their peers or servers in the wireline network through the AP. As well, all types of traffic destined for wireless stations are also via the AP from the wireline network. The upstream and the downstream traffic loads for all types of services are assumed to be the same.  We assume that the hotspot consists of 6 voice, 2 video, and n FTP client stations. Since 802.1 le E D C F is designed to be backwards compatible to 802.11 DCF, we assume that half of the FTP clients are using the 802.11 D C F while the other half are using the new 802.1 le E D C F protocol with the B E B algorithm with PF value of 2.0.  We simulate the  hotspot scenario to compare the A D B algorithm for real-time traffic against the generalized BEB algorithm with PF values of 1.5 and 2.0 for real-time traffic. Again, best-effort FTP data traffic employs the conventional BEB algorithm with PF = 2.0.  5.2.2  Simulation Results from the Hotspot Scenario This section presents results from simulations of the hotspot scenario. Voice packet  delay is shown in Figure 5.10.  Uplink and downlink voice packet jitter is shown in Figure  5.11a and 5.11b respectively, where jitter is defined as the delay variance. The uplink and downlink voice drop rate is shown in Figure 5.12a and 5.12b, respectively. The lifetime of a voice packet is assumed to be 25 ms.  54  0.02  10  12  14  16  18  20  number of FTP clients •E D C F (PF 2.0)-Uplink  - • — A D B - Uplink - A — E D C F (PF=1.5)-Uplink •• -  —o—  E D C F (PF=2.0) - D o w n l i n k — A -  A D B - Downlink E D C F (PF=1.5) - Downlink  Figure 5.10. Uplink and Downlink voice packet delay in a hotspot  55  22  1.20E-03  10  12  14  16  18  number of FTP clients • E D C F (PF=2.0) - Downlink  - A D B - Downlink -EDCF  (PF=1.5)-Downlink  (b)  Figure 5.11. (a) Uplink voice packet j itter in a hotspot. (b) Downlink voice packet jitter in a hotspot  56  10  12  14  18  16  20  22  number of FTP clients -ADB  - Uplink  • E D C F (PF=2.0) - Uplink  •EDCF  (PF=1.5)-Uplink  (a)  10  12  14  16  18  20  number of FTP clients • E D C F (PF=2.0) - Downlink  - A D B - Downlink -EDCF  (PF=1.5)-Downlink  (b)  Figure 5.12. (a) Uplink voice packet drop rate in a hotspot (b) Downlink voice packet drop rate in a hotspot  57  22  Figure 5.13 shows video packet delay.  Figure 5.14a and 5.14b show uplink and  downlink video packet jitter. Figure 5.15a and 5.15b show uplink and downlink video packet drop rates.  0.25  10  12  14  16  18  20  number o f FTP clients • E D C F (PF=2.0) - Uplink  - A D B - Uplink — A — E D C F — • -  (PF=1.5)-Uplink  — o—  E D C F (PF=2.0) - Downlink — A -  A D B - Downlink E D C F (PF=1.5) - Downlink  Figure 5.13. Video packet delay in a hotspot  58  22  0.009  10  12  14  18  16  20  22  number of FTP stations - A D B - Uplink  • E D C F (PF=1.5) - Uplink  • E D C F (PF=2.0) - Uplink  (a)  12  14  16  22  number of FTP clients -ADB -EDCF  •E D C F (PF=2.0) - Downlink  - Downlink (PF=1.5)-Downlink  (b) Figure 5.14. (a) Uplink video packet jitter in a hotspot (b) Downlink video packet jitter in a hotspot  59  12  10  14  16  18  20  22  number of FTP clients -ADB  • E D C F (PF=2.0) - Uplink  - Uplink  •EDCF  (PF=1.5)-Uplink  (a)  number of FTP clients - A D B - Downlink -EDCF  • E D C F (PF=2.0) - Downlink  (PF=1.5)-Downlink  (b)  Figure 5.15. (a) Uplink video packet drop rate in a hotspot (b) Downlink video packet drop rate in a hotspot  60  The main performance characteristic of data traffic is measured by throughput. The total throughput of the FTP data traffic in the hotspot network is shown in Figure 5.16.  •3-  700000  ~ 3  600000  £  500000  o  400000  Z  300000  1 0  200000  t co  o  100000 0  -I  ,  ,  ,  ,  ,  ,  ,  1  6  8  10  12  14  16  18  20  22  number of FTP clients —•—ADB - » - E D C F  (PF=2.0) - A - E D C F  (PF=1.5)  Figure 5.16. Total FTP data throughput in a hotspot  5.2.3  Discussions of the Hotspot Results Figures 5.10, 5.11 and 5.12 show that, relative to E D C F without A D B , voice packet  delay, jitter and drop rate improve significantly as the number of FTP stations increases when E D C F incorporates A D B .  Figures 5.13, 5.14 and 5.15 demonstrate that with A D B , video  packet delay, jitter and drop rate all improve considerably when many best-effort FTP stations are active.  61  Like the simulation results from the ad-hoc scenario, those from the hotspot scenario indicate that when the A D B retransmission algorithm is employed in the new 802.1 l e E D C F protocol, the delay, jitter, and packet drop rate o f the voice and video traffic is reduced without sacrificing the throughput o f best-effort F T P traffic (see Figure 5.16). The results suggest that A D B enhances Q o S performance o f E D C F in the hotspot.  The  EDCF  protocol  provides service differentiation,  which  is an  important  improvement over 802.11 D C F . E D C F presents delay asymmetry, giving an advantage to uplink transmission as shown in Figures 5.10 and 5.13.  This is because the aggregate  downlink real-time traffic sent by the A P must complete for the channel in equal terms with all the stations that want to transmit in the uplink direction.  The A D B algorithm reduces delay asymmetry by giving priority to real-time packets with high age values. Downlink real-time traffic having long queuing delay w i l l be assigned to use small P F value as adjustment to their C W sizes.  5.3  Conclusion from the Simulations Results  Results from both the ad-hoc and hotspot  scenarios suggest that  the A D B  retransmission algorithm used in the new 802.1 l e E D C F protocol offers improvement, relative to that from use o f the B E B algorithm (with real-time traffic) in Q o S differentiation, under a wide range o f traffic loads, in both ad-hoc networks and in hotspots.  The  performance o f the real-time traffic is enhanced without reducing the throughput o f the besteffort traffic.  62  Service differentiation becomes more noticeable and more pronounced as the number n of best-effort stations increases. A D B can also reduce delay asymmetry at the hotspot. The difference between the uplink and the downlink delay is reduced significantly when the A D B algorithm is employed.  There is a penalty for using A D B to reduce delay, jitter, and packet drop rate of realtime traffic while maintaining throughput levels of best-effort data traffic. The best-effort traffic delay and jitter increase, in some cases substantially, when the A D B algorithm is employed.  However such delay and jitter increase, not presented here, is not regarded as  problematic for best-effort data traffic.  63  Chapter 6 Summary and Conclusions The increasing use o f wireless and Internet communications has created a strong demand for public Internet access over 802.11 wireless L A N s . I E T F is currently working on service differentiation at the IP layer, but the result is sub-optimal without lower layers' support. Since 802.11 wireless L A N s appear everywhere, it is essential to focus on service differentiation support at the 802.11 M A C layer.  To improve the current 802.11 M A C protocol to support applications with Q o S requirements, the I E E E 802.11 Task Group E was formed and is defining Q o S enhancements for 802.11 M A C . The 802.1 l e draft introduces the E D C F protocol, which is a prioritization enhancement o f the 802.11 D C F protocol.  In the current version o f  E D C F , the B E B algorithm is employed i n E D C F to resolve collisions; however B E B causes long delay and large jitter that are unfavorable for real-time packets with timebounded requirements.  This thesis is an extension to the current work on providing QoS enhancements in the 802.11 M A C layer studied by I E E E 802.11 Task Group E . In particular, we focus on the improvement o f the 802.1 l e E D C F protocol by proposing our A D B retransmission algorithm.  64  6.1  Summary of the Work The primary contribution of this thesis is our proposal, analysis, and evaluation of  the A D B retransmission algorithm that can alleviate delay, jitter, and drop rate for realtime traffic with time-bounded requirements without reducing the throughput of besteffort data traffic.  The A D B algorithm is used together with the B E B algorithm in  802.11 wireless LANs.  The A D B algorithm is for high priority real-time traffic, while  the BEB algorithm is for best-effort data traffic.  The A D B retransmission mechanism is easy to implement.  It requires minor  modifications in the computation of C W thereby minimizing the migration effort from the new 802.1 le EDCF protocol. Our A D B retransmission strategy provides backward compatibility to the 802.11 DCF protocol. The implementation of the A D B algorithm is relatively simple and is practical with current software and hardware technologies.  OPNET simulation models are built to study the performance of the new 802.11 E D C F protocol with the proposed A D B retransmission algorithm in two typical environments which are modeled by two simulation scenarios, namely an ah-hoc network and a hotspot scenarios.  The results from both scenarios indicate that using A D B in  EDCF provides low delay, jitter and drop rate for real-time traffic. The delay asymmetry which exists without A D B is reduced significantly in the hotspot environment when A D B is employed. The improvements are more noticeable and more pronounced as the data traffic load increases.  65  In conclusion, A D B is a useful retransmission algorithm for the new 802.1 l e E D C F protocol.  Since A D B can be implemented by changing the software without  alternating the existing hardware, the cost of implementation is considered to be relatively low.  6.2  Future Work To further extend the work of this thesis, the following possible directions for  future research are suggested.  1. In our work, it was assumed that all packets are transmitted at the same rate.  In  reality, some stations may transmit faster than others depending on the wireless link quality. Simulation of the proposed A D B algorithm in a noisy environment is desirable for extending validation of the results presented here.  2.  Admission control is a key aspect for the real4ime mechanism to work well, since such control limits the amount of real-time traffic admitted to 802.11 wireless LANs.  A n exact design of an admission control scheme for the new 802.11  E D C F protocol with the proposed A D B algorithm requires further investigation.  3.  The proposed A D B retransmission algorithm can be applied to the C S M A / C D protocol to support QoS differentiation in 802.3 Ethernet. It would be interesting to study the performance of the 802.3 C S M A / C D protocol with A D B .  66  References  [1]  William Stalling, "IEEE 802.11: Moving closer to practical wireless LANs", IT Professional, vol. 3, issue 3, pp. 17-23, May-June 2001.  [2]  Jim Geier, Wireless LANs, Second Edition, Indianapolis, Indiana: Sams Publishing, 2002.  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