Performance of Voice and Data Transmission Using the IEEE MAC Protocol

Transcription

1 Performance of Voice and Data Transmission Using the IEEE MAC Protocol by Jessica M. Yeh Submitted to the Department of Electrical Engineering and Computer Science in Partial Fulfillment of the Requirements for the Degree of Master of Engineering in Electrical Engineering and Computer Science at the MASSACHUSETTS INSTITUTE OF TECHNOLOGY June 2002 c Jessica M. Yeh, MMII. All rights reserved. The author hereby grants to M.I.T. permission to reproduce and distribute publicly paper and electronic copies of this thesis and to grant others the right to do so. Author... Department of Electrical Engineering and Computer Science May 24, 2002 Certified by... Jon Anderson Senior Staff Systems Engineer VI-A Company Thesis Supervisor Certified by... Professor Vincent W.S. Chan Director, EECS Laboratory for Information and Decision Systems M.I.T. Thesis Supervisor Accepted by Arthur C. Smith Chairman, Department Committee on Graduate Theses

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3 Performance of Voice and Data Transmission Using the IEEE MAC Protocol by Jessica M. Yeh Submitted to the Department of Electrical Engineering and Computer Science on May 24, 2002, in Partial Fulfillment of the Requirements for the Degree of Master of Engineering in Electrical Engineering and Computer Science Abstract The IEEE wireless LAN standard attempts to provide high throughput and reliable data delivery for stations transmitting over a lossy, wireless medium. To efficiently allocate resources for bursty sources, the Medium Access Control (MAC) sublayer uses a type of Carrier Sense Multiple Access with Collision Avoidance (CSMA/CA) protocol called the Distributed Coordination Function (DCF). The MAC protocol also includes an optional polling scheme called the Point Coordination Function (PCF) to deliver near-isochronous service to stations. This thesis analyzes the performance of these two medium access mechanisms under real-time voice and asynchronous data transmissions. Using analytical and simulative methods, the efficiency and capacity of the protocol is determined for each type of traffic individually, as well as for a traffic mix of the two types. It is shown that the upper bound of data efficiency for DCF is 65.43% percent when transmitting maximumsized IP packets at 11 Mbps. Furthermore, due to the difference in packet size of the two traffic types, for each additional GSM voice call (approximately 11 kbps including voice activity) to be supported using DCF, the non-real-time traffic load must decrease by approximately 250 kbps. Voice receives very little real-time Quality of Service (QoS) when using DCF to contend with constantly sending data stations. In order for to provide real-time QoS for voice packets despite all levels of asynchronous traffic data load, the PCF mechanism can be used. By only using PCF for voice traffic, voice packets will always take priority over asynchronous data packets and receive the required real-time QoS. VI-A Company Thesis Supervisor: Jon Anderson Title: Senior Staff Systems Engineer M.I.T. Thesis Supervisor: Professor Vincent W.S. Chan Title: Director, EECS Laboratory for Information and Decision Systems 3

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5 Acknowledgments I would like to express my appreciation to my thesis advisors, Jon Anderson at Qualcomm, who supported me through all logistical obstacles, for his mentorship and motivation, and Professor Vincent W.S. Chan at M.I.T. for his continued support and advice, despite project changes. My appreciation also goes to Rajiv Vijayan and his group at Qualcomm in San Diego for giving me the opportunity to work in such a exciting field. Special thanks to Bruce Collins for providing guidance and direction for the project. I would also like to acknowledge Jeanne Clark at Qualcomm for all her efforts in setting up OPNET, and Mark Guzzi at Qualcomm for his helpful comments and suggestions in defining the project. Finally, I thank my family and friends for their continued support and encouragement throughout my work on this project. 5

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7 Contents 1 Overview Background WirelessLocalAreaNetworks Multiaccess Schemes QualityofService ProjectOverview RelatedStudies ProjectObjectives IntroductiontotheFollowingChapters IEEE Standard Architecture FrameFormat TheMACProtocol Medium Access Mechanisms TimingIntervals Distributed Coordination Function (DCF) Point Coordination Function (PCF) Concurrent Operation of DCF and PCF BasicFrameExchange Theoretical Analysis AnalysisoftheDCF

8 3.1.1 DataEfficiency Theoretical Upper Bound of Network Throughput CalculatedValues DCF Contention Among Several Stations AnalysisofthePCF DataEfficiency CalculatedValues EffectsofTrafficActivity ComparisonofDCFandPCF Simulation Setup Assumptions DefaultSimulationParameters RadioChannelModel AsynchronousDataTrafficModel DataPacketSize DataPacketDestination DataPerformanceMetrics AsynchronousDataUserTypes PoissonArrivalUser ConstantlySendingUser Real-timeTrafficModel Voice Packet Stream Characteristics VoicePacketDestination VoicePerformanceMetrics BSSOperationScenarios Asynchronous Data Transmission Using DCF Real-time Transmission Using DCF Supporting Two Types of Traffic Using Only DCF Real-timeTrafficUsingMostlyPCF

9 4.7.5 Supporting Two QoS Using DCF and PCF Simulation Results and Analysis Asynchronous Data Transmission Using DCF Varying Data Load with a Constant Number of Users Constant Data Load with a Varying Number of Users Real-timeVoiceTransmissionUsingDCF Supporting Two Types of Traffic With DCF Voice Traffic Contending With Poisson Arrival Users Voice Contending With Constantly Sending Users Real-timeVoiceTrafficUsingMostlyPCF Supporting Two QoS Using DCF and PCF Enhancements for Service Differentiation InterframeSpace MinimumContentionWindowSize SimulationSetup Results Summary and Conclusion 89 A Calculating Backoff 93 9

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11 List of Figures Architecture IEEEMACFrameFormat Basic Access Mechanism BinaryExponentialBackoff PCF Operation during the Contention Free Period (CFP) DCFandPCFSuperframeStructure BasicFrameExchange Frame Exchange Using RTS/CTS TheoreticalAverageFinalBackoff Average Throughput using Theoretical Average Final Backoff Data Efficiency of PCF When Transmitting 2304-byte Packets Data Efficiency of PCF When Transmitting 32.5-byte Packets Data Efficiencies of DCF and PCF, Varying the Number of Stations Average Data Throughput for Poisson Arrival Stations Using DCF Average Data Packet Delay for Poisson Arrival Stations Using DCF Average Throughput for Constantly Sending Stations Using DCF Average Final Continuous Backoff for Constantly Sending Stations UsingDCF Average Number of Retransmissions at One Station for Constantly SendingStationsUsingDCF CDFofVoiceUsingDCF,GSMencoding CDF of Voice Using DCF, G.711 encoding

12 5-8 Probability of Voice Packets Meeting Delay Threshold of 25 msec CDF of 1 Voice Call Contending With Poisson Arrival Data Stations Varying the Mean Data Load, GSM encoding CDF of 1 Voice Call Contending With Poisson Arrival Data Stations Varying the Mean Data Load, G.711 encoding CDF of 5 Voice Calls Contending With Poisson Arrival Data Stations Varying the Mean Data Load, GSM encoding CDF of 5 Voice Calls Contending With Poisson Arrival Data Stations Varying the Mean Data Load, G.711 encoding CDF of 9 Voice Calls Contending With Poisson Arrival Data Stations Varying the Mean Data Load, GSM encoding CDF of 9 Voice Calls Contending With Poisson Arrival Data Stations Varying the Mean Data Load, G.711 encoding Probability of Voice Packets Meeting Delay Requirement of 25 msec whencontendingwithpoissonarrivalusers CDF of 1 Voice Call Contending With Constantly Sending Data Stations,GSMencoding CDF of 1 Voice Call Contending With Constantly Sending Data Stations,G.711encoding CDF of 5 Voice Calls Contending With Constantly Sending Data Stations,GSMencoding CDF of 5 Voice Calls Contending With Constantly Sending Data Stations,G.711encoding CDF of 9 Voice Calls Contending With Constantly Sending Data Stations,GSMencoding CDF of 9 Voice Calls Contending With Constantly Sending Data Stations,G.711encoding Probability of Voice Packets Meeting Delay Requirement of 25 msec when Contending with Constantly Sending Users Delay Characteristics of Voice Calls Using PCF, GSM encoding

13 5-24 CDF of Voice Calls Using PCF, GSM encoding Percent of High Priority Throughput, 1 high priority station and 1 low prioritystation Percent of High Priority Throughput, 1 high priority station and 2 low prioritystations

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15 List of Tables 3.1 Protocol Parameters for DCF using a DSSS Physical Layer Data Efficiencies and Effective Data Throughput for Sending 2304-byte PacketsUsingDCF Data Efficiencies and Effective Data Throughput for Sending 1500-byte PacketsUsingDCF Data Efficiencies and Effective Data Throughput for Sending 32.5-byte PacketsUsingDCF Protocol Parameters for PCF using a DSSS Physical Layer Data Efficiencies and Effective Data Throughput for PCF: 2304-byte Packets,1pollperCFP Data Efficiencies and Effective Data Throughput for PCF: 32.5-byte Packets,1pollperCFP Data Efficiencies and Effective Data Throughput for PCF: 32.5-byte Packets with Varying Traffic Activity, 1 poll per CFP SimulatedProtocolParameters Poisson Arrival Data Stations and Voice Stations Both Using DCF Number of Constantly Sending Data Stations Contending With Voice Stations Without Breaking the Voice Call, All Using DCF Asynchronous Data Throughput Using DCF with Voice Using PCF

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17 Acronyms A definition of the listed acronym can be found on the page(s) listed in parentheses. ACK acknowledgment (35) AP access point (28) BSS basic service set (28) CFP contention-free period (33) CP contention period (31) CSMA/CA carrier sense multiple access with collision avoidance (29) CTS clear to send (35) CW contention window (32) DCF distributed coordination function (31) DIFS distributed (coordination function) interframe space (30, 31) DS distribution system (29) DSSS direct sequence spread spectrum (27) EIFS extended interframe space (30) ESS extended service set (29) FHSS frequency-hopping spread spectrum (27) LAN local area network (19) MAC medium access control (21) MPDU MAC protocol data unit (38) NAV network allocation vector (30) PC point coordinator (33) PCF point coordination function (33) 17

18 PHY physical (layer) (27) PIFS point (coordination function) interframe space (30, 33) RTS request to send (35) SIFS short interframe space (30) TC traffic class (85) 18

19 Chapter 1 Overview Network computing provide an abundance of resources to an end user on a single computer. Hardware and software applications can easily be shared among multiple personal computers. Other applications such as FTP can enable transfers of files through the network between remotely located machines. Furthermore, the rapid growth of the World Wide Web in recent years brings a wealth of information and services, all conveniently accessible through an Internet connection from a computer in the home. As we become accustomed to the benefits provided by computer networks, there is a growing desire for continuous network connection. Our busy lives demand portable devices that can keep us connected throughout the mobility of our daily lives without the hassles of cables and wires. Wireless local area networks (LANs) provide much of the desired flexible functionality. Because they do not require an existing wired infrastructure, wireless LANs can be easily created without the need for extensive cable installation or other changes of the existing network. Furthermore, with little difficulties, they can be modified or replaced as needed, providing a convenient possibility for building simple, temporary networks. Users with portable devices may travel anywhere within the basic service area, all the while maintaining a connection to the LAN. Thus, wireless LANs can easily function as an extension of a wired LAN giving additional flexibility to the existing structure. This thesis studies the performance of the IEEE wireless LAN protocol. 19

20 The two medium access mechanisms of the protocol are analyzed under real-time voice and asynchronous data loading to determine the effectiveness of the protocol in offering Quality of Service for real-time traffic. 1.1 Background Wireless Local Area Networks To an end-user, wireless networks should function almost identically to wired networks. Wireless LANs must have a method of concealing the nature of the physical network and seamlessly allowing for mobility. Especially when contention for limited media resources occurs among several stations, the wireless LAN must be able to fairly and efficiently allocate these resources. However, problems often arise with the use of a wireless environment, and these issues are resolved in the medium access protocol. For economic feasibility, wireless LAN devices cannot simultaneously listen to the medium while transmitting because they usually have only one antenna available for both sending and receiving. Thus, collision detection algorithms that continuously monitor the medium, such as those used in Ethernet, are much more difficult to implement. When switching between the circuits responsible for sending and receiving, the interface will not be able to perform either task for a certain period of time. This so-called Rx/Tx-turnaround-time places restrictions on the speed of exchange possible in the medium access protocol [10]. Furthermore, due to interference among co-located wireless LANs, the wireless channel experiences higher error rates compared to those of wired channels [10]. Thus, dropped data cannot always simply be attributed to congestion in the transmission medium. Due to the limits in the range of signal propagation, wireless LANs encounter another issue known as the hidden-node problem. Station A may not be within receiving range of a currently sending station, B, and thus will consider the medium to be idle. Station A may, however, be within range of the receiving station C of the 20

21 current transmission from B. Any attempts to initiate a new transmission from A to C may corrupt signals from both A and B. Having no means of collision detection, the current senders A and B would both continue to transmit, resulting in a lot of wasted bandwidth. Wireless LANs have far less bandwidth available than wired LANs. Current commercial wireless LAN products support data rates up to only 11 Mbps. Furthermore, the Federal Communications Commission (FCC) allocates a relatively small amount of bandwidth for the use by wireless LANs. Because bandwidth enhancements are difficult to achieve in wireless LANs [5], this scarce resource must be used efficiently Multiaccess Schemes When several stations share one single medium for transmission, a protocol is needed to control access to this resource. Because stations operate independently, a station will not know when another station needs to use the medium to transmit a packet. By restricting access with a specified protocol, collisions of multiple stations simultaneously attempting to transmit can be reduced. In addition, various techniques can be used to ensure the intended data is transmitted with minimal errors. For these multiaccess networks, the mechanism that governs access to the common medium resides in the Medium Access Control (MAC) Layer, the lower sublayer of the International Standards Organization (ISO) Open System Interconnection (OSI) Basic Reference Model s Data Link Control Layer (Layer 2). MAC protocols typically can be categorized into several categories: fixed assignment, random access, and dynamic demand assignment. Fixed assignment protocols such as Time Division Multiple Access (TDMA), Frequency Division Multiple Access (FDMA), and Code Division Multiple Access (CDMA) devote a fixed amount of resources to each user of the channel. However, these protocols often suffer from inefficient use of the resources. For a network of bursty sources, to accommodate the worst-case traffic load, much of the allocated resources would be wasted during periods of inactivity. Random access protocols such as ALOHA, Carrier Sense Multiple Access with 21

22 Collision Detection (CSMA/CD), and Carrier Sense Multiple Access with Collision Avoidance (CSMA/CA) rely on stations contending for control of the medium through stochastic means. This enables a network to fairly allocate resources as needed by each station s traffic load. These distributed medium access mechanisms require little coordination and are effective for low or medium load conditions. However, as the traffic load grows, the probability of collision during channel access contention increases, resulting in longer packet delays and throughput well less than 100% [5]. Dynamic demand assignment protocols such as Token Ring, Packet Reservation Multiple Access (PRMA), and Demand Assignment Multiple Access (DAMA) attempt to combine the deterministic behavior of fixed assignment with the flexibility of random assignment. With the expense of more coordination, better performance under higher traffic loads can be achieved. For wireless packet data networks, fixed assignment protocols seem unsuitable because they lack the adaptability in allocating resources and allowing frequent configuration changes. Demand assignment protocols are often difficult to implement for wireless networks due to some of the requirements to accommodate mobility. For example, token-based schemes rely on knowledge of the current network topology so each station knows which stations are its current neighbors, which may be a tedious task to maintain in mobile configurations. Wireless networks need a protocol to accommodate the possibility of a constantly changing network topology. Thus, random assignment methods seem the ideal choice to allow for free movement by the mobile device. However, the tradeoff for flexibility is a non-deterministic behavior that cannot always guarantee support for a desired Quality of Service Quality of Service Quality of Service (QoS) refers to the ability of a network to effectively provide a certain level of support for selected classes of network traffic. With QoS, the LAN features a more predictable network service by supporting dedicated bandwidth, improving loss characteristics, and setting traffic priorities across the network. In this way, QoS provides more guarantees for transmissions across the network. 22

23 Quantitatively, QoS can be described with parameters such as frame error rate, latency, jitter, and capacity. Frame error rate is the amount of frames lost or corrupted en route through the network. Latency describes the delay experienced by the traffic as it travels across the network, and jitter represents the variation of delay experienced by different frames in a stream of traffic. Capacity is the amount of useable bandwidth available for the session to transmit data. Some guarantees regarding ordered delivery of packets may also be assumed for a certain QoS. With networks equipped to support different levels of QoS, various types of traffic can experience different forms of reliable delivery over the same network. For example, data applications and other asynchronous types of data require bandwidth for efficient transfer of large amounts of data with little packet errors, while being able to tolerate latency and jitter. On the other hand, real-time data such as voice or video need a dedicated amount of bandwidth with short latency, low jitter, little packet loss, but not necessarily completely error-free transmission. With well-designed QoS support, a network can allocate resources to perform a high-quality voice or other time-critical transmission while maintaining efficient asynchronous data traffic flow. 1.2 Project Overview For this project, the performance of the IEEE MAC protocol in offering QoS for various types of traffic is evaluated. Specifically, the two medium access mechanisms, the random assignment Distributed Coordination Function (DCF) and the dynamic demand assignment Point Coordination Function (PCF), are analyzed Related Studies Previous studies have been conducted to model the performance of the IEEE MAC protocol. Several papers investigate the performance of the Distributed Coordination Function of the protocol in an ad hoc network under asynchronous data traffic [3, 10]. These studies evaluate how certain tunable parameters of the standard such as packet size, Request To Send/Clear To Send (RTS/CTS) threshold, and frag- 23

24 mentation threshold affect the network throughput and delay. Through simulation, it is shown that with low channel error rates, a reasonably high channel efficiency can be achieved. A study by Kopsel et al. [5] compares the performance of the Distributed Coordination Function with the Point Coordination Function under real-time traffic requirements. They modeled the load as a dual-source mix of voice and asynchronous data traffic and determined that DCF works well under low load conditions, but experiences throughput deterioration under high load conditions due to the increased time needed to contend for the channel. Meanwhile, the centralized-control protocol, PCF, works well under high load scenarios by optimizing channel bandwidth utilization and decreasing packet wait-time, though there is often high overhead due to unsuccessful polling attempts Project Objectives This project investigates the performance of real-time traffic over DCF and PCF of the IEEE MAC protocol. The primary objectives of this study include examining the throughput and capacity performance of the MAC layer with respect to the following traffic loads: asynchronous data users, users demanding a real-time QoS, and a combination of these two user types. The ability of the MAC protocol to deliver the QoS required of real-time traffic, as well as the degradation to asynchronous data throughput caused by supporting real-time traffic are studied. 1.3 Introduction to the Following Chapters This chapter has introduced some of the issues of consideration in designing wireless LAN protocols. Different types of multiaccess schemes have also been described. Fixed assignment protocols guarantee a fixed amount of resources to each user of the channel, but may suffer from inefficiencies due to resources allocated to idle users. Random access protocols rely on contention for access among many users to allocate resources as needed by each station s specific traffic load, but may experience packet 24

25 collisions, especially at high traffic loads. Dynamic demand assignment protocols combine the advantages of the two above protocol types, and with more coordination, may achieve better performance under high traffic loads. With these issues in mind, the different multiaccess schemes of the IEEE MAC protocol are evaluated. Chapter 2 gives a brief summary of the IEEE wireless LAN standard. The basic architectural components of are introduced, and the MAC protocol is explained. This chapter also describes the different processes by which stations may access the wireless medium and introduces the sequence of frame exchanges which may occur between stations. Chapter 3 presents a theoretical analysis of both access mechanisms of the MAC protocol. The overhead of the DCF mechanism is analyzed in terms of data efficiency, and an upper bound on network throughput is derived. Similar values are also calculated for the PCF mechanism, and the results are compared. The simulation study is introduced in Chapter 4. The scope and design of the simulation is presented, and relevant assumptions are explained. The basic architecture of the network being simulated is described, as well as the simulated user types at each station. This chapter presents the different operation scenarios of the network, and describes the traffic models used. Metrics used to evaluate performance are also given. Chapter 5 presents results of the simulation study. Data collected from the simulations is analyzed, and conclusions about the medium access protocols are drawn. Chapter 6 presents additional mechanisms under consideration for enhancing the current standard protocol with differentiation schemes. The methods of providing service differentiation are described, and the results from simulating these mechanisms are presented. Chapter 7 concludes with a brief summary of the results of the simulation study and suggestions for further research. 25

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27 Chapter 2 IEEE Standard The standard was devised under the IEEE 802 family of standards for local and metropolitan area networks, which also includes common standards such as Ethernet (802.3) and Token Ring (802.5). Similar to the other standards in the family, the standard pertains to the Physical and Data Link layers as defined by the ISO/OSI Basic Reference Model [1]. Defined in is the Medium Access Control (MAC) layer, MAC management and protocol services, and three physical layers (PHY). The physical layers include an infrared baseband PHY, a frequency hopping spread spectrum (FHSS) PHY, and a direct sequence spread spectrum (DSSS) PHY. The goal of the task group was to devise a standard to describe a wireless LAN that delivers high throughput, reliable data delivery, and continuous network connections, resembling characteristics previously only available for wired networks [6]. Currently, there have been two flavors of the standard released a describes requirements for a high-speed physical layer in the 5 GHz band that offers transmission rates up to 54 Mbps, while b describes a high-speed physical layer in the 2.4 GHz band, offering rates up to 11 Mbps. 27

28 2.1 Architecture The architecture is comprised of several components: the Station (STA), the Access Point (AP), the wireless medium, the Basic Service Set (BSS), the Distribution System (DS), and the Extended Service Set (ESS). BSS AP ESS DS AP STA_1... STA_n BSS AP BSS Figure 2-1: Architecture The Station is the component that connects to the wireless medium, typically a PC or a PCI card. The BSS is the basic network architectural component that is composed of two or more stations communicating with each other. If the stations in a BSS communicate directly with one another, they are said to be operating in ad hoc mode. When they communicate through a mediating station, they are said to be in infrastructure mode, with the mediator being known as the AP. The AP is a 28

29 specialized station that can also connect a BSS to another wired or wireless network. The means by which APs communicate with each other is through an abstract medium known as the DS. This can be either a wired network such as Ethernet, or another wireless network. When several different BSSs comprise a network, they, together with the DS, form an ESS. 2.2 Frame Format The general MAC frame format specifies a set of fields that are present in a fixed order in all MAC frames. The general MAC frame format is shown in Figure 2-2. With the exception of the Address 4 field, all depicted fields occur in all MAC data frames. The Address 4 field is only used if the wireless network is being used to implement the DS. Other fields, such as Address 2, Address 3, Sequence Control, Address 4, and Frame Body, may be omitted in certain other frame types. (Please reference [1, 6] for definitions of each field and detailed descriptions of the formats of each individual frame type.) MAC Header Frame Control Duration/ ID Address 1 Address 2 Address 3 Sequence Control Address 4 Frame Body FCS Octets: Figure 2-2: IEEE MAC Frame Format 2.3 The MAC Protocol The IEEE MAC layer uses a type of random assignment protocol known as Carrier Sense Multiple Access with Collision Avoidance (CSMA/CA) with binary exponential backoff. However, it also provides an optional demand assignment scheme 29

30 in order to deliver near-isochronous service to stations [6]. Both a and b use the same MAC protocol. In CSMA, the physical layer of a station will perform carrier sensing by listening to the medium to ensure that another transmission is not already in progress before beginning its own transmission. In addition to the physical carrier sense mechanism provided by the physical layer, also uses a virtual mechanism in an effort to avoid collisions on the wireless medium. A value in the network allocation vector (NAV) maintained by the MAC in each station indicates to the station how much longer the medium will be busy. This value is updated from duration values found in all transmitted frames. Thus, each station decodes the MAC header of every frame it hears to keep track of network activity. 2.4 Medium Access Mechanisms The protocol describes two medium access mechanisms: the random access Distributed Coordination Function (DCF) and the demand assignment Point Coordination Function (PCF). Five timing intervals that control access to the shared wireless medium are used to implement the two access mechanisms Timing Intervals Figure 2-3 shows the relative lengths of the timing intervals. The shortest interval is the short interframe space (SIFS), which is the separation of frames within a transmission sequence of the frame exchange protocol. A slightly longer interval is the slot time. The PCF interframe space (PIFS) is equal to one SIFS plus one slot time, and the DCF interframe space (DIFS) is equal to one SIFS plus two slot times. The extended interframe space (EIFS) is much longer than the DIFS, and is used to allow stations to regain timing synchronization with the rest of the network when a transmission is received in error. The duration of the basic timing intervals are specified according to the particular physical layer being used. 30

31 Backoff Previous Transmission Next Transmission Short InterFrame Space (SIFS) PCF InterFrame Space (PIFS) Slot Time DCF InterFrame Space (DIFS) Extended InterFrame Space (EIFS) Figure 2-3: Basic Access Mechanism Distributed Coordination Function (DCF) The DCF is the basic mechanism that controls access to the wireless medium. All stations are required to support DCF services. The period during which the DCF operates is referred to as the Contention Period (CP). After receiving a request for transmission from higher layer protocols, the MAC will check both physical and virtual carrier sense mechanisms. Once the medium is determined to be idle by both sensing mechanisms simultaneously for an interval of DIFS (or EIFS if the previous transmission contained errors), the MAC may begin transmitting the frame. If the medium is determined to be in use during the DIFS-interval, the MAC will increment the retry counter associated with that frame and defer until an idle DIFS-interval to begin backing off. Transmission of the frame can begin only when the backoff timer has expired Backoff In order to prevent stations deferring to a transmission from all attempting to send data immediately following completion of the current transmission, the protocol requires stations to perform a binary exponential backoff. After sensing that the medium is idle for a DIFS-interval, a station wishing to transmit a frame will randomly select a deferral value to use as its backoff timer. The backoff timer is selected 31

32 DIFS DIFS DIFS Contention Window Contention Window Station 1 Station 2 Defer Station 3 Defer Station 4 Defer = Transmission = Backoff = Remaining Backoff Figure 2-4: Binary Exponential Backoff from a uniform distribution over a range known as the contention window (CW). This timer value is decremented by one for each slot time the MAC determines the medium to be idle following the idle DIFS-interval. Should the medium become busy during backoff, the backoff timer will suspend countdown. Once the medium again becomes idle for a DIFS-interval, the station will resume counting down the backoff timer from the value when it was last suspended. The station only transmits the frame when its backoff timer expires. Figure 2-4 shows an example of how the backoff procedure works. To prevent one station with a lot of traffic from consuming all the bandwidth of the wireless medium, after a successful transmission, the station must perform backoff using a minimum-sized contention window before attempting a subsequent transmission. If the transmission is unsuccessful (i.e. no ACK is received), a collision is assumed 32

33 to have occurred (regardless of whether this actually happened). The contention window size doubles (unless it has already reached maximum size), a deferral value is selected using the new contention window, and the backoff timer begins counting down once more. The process continues until the transmission is successful, the maximum specified retry limit is reached, or the transmission is cancelled by higher layer protocols. When a successful transmission is completed, the contention window returns to its minimum size. The specific physical layer being used determines the minimum and maximum size of the contention window. Due to the combination of contention and backoff employed by DCF, stations may experience extremely long wait-times for access to the medium. This possibly long delays as well as wide variation in delay times may be detrimental to real-time traffic. Thus, to support time-bounded traffic, the MAC protocol also includes a centralized mode, the Point Coordination Function (PCF), that is governed by a demand assignment scheme Point Coordination Function (PCF) The PCF is an optional mechanism that uses a poll and response method to eliminate contention for the medium. In this centrally controlled mechanism, the point coordinator (PC) located in the AP controls access to the wireless medium. The PC gains access to the medium using procedures similar to those used in DCF. However, instead of waiting for a DIFS-interval, it is only required to wait a PIFS-interval before determining the medium is idle and taking control of the medium. Once the PC has acquired the medium, it sends a beacon frame notifying stations of the beginning of the period of PCF operation known as the Contention-Free Period (CFP). The beacon contains the maximum expected duration of the CFP, which stations use to update their NAVs. During the CFP, the PC delivers frames to stations while also individually polling stations that have previously registered on the polling list requesting contention-free service. Each station can send one data frame for each CF-Poll received. By setting an appropriate CFP repetition interval, this mechanism can guarantee a bounded 33

34 Contention Free Period (CFP) Contention Period (CP) SIFS SIFS PIFS SIFS AP Station Beacon Data+Poll ACK+Data ACK+Poll Poll Data ;;; Backoff ACK+ CFEnd PIFS SIFS SIFS DIFS NAV NAV Reset CFP Maximum Duration Figure 2-5: PCF Operation during the Contention Free Period (CFP) delay for transmission of packets arriving at stations that have requested this service. To maintain control of the medium during CFP, the PC ensures that the interval between frames is no longer than PIFS. If the PC does not receive a response to a data transmission or a CF-Poll within a period of SIFS, it will transmit its next frame before a PIFS concludes. Figure 2-5 depicts an example of possible frame transmissions during a CFP. The end of the CFP is announced when the PC sends a contention-free-end (CF-End) frame. With this frame, stations reset their NAVs and may begin competing for access to the medium under normal DCF methods Concurrent Operation of DCF and PCF Because the PCF mechanism uses DCF methods to obtain control of the medium, it is not required that all stations support PCF services. The PC uses the PIFS interval to operate concurrently with the DCF and gain access to the medium to begin the PCF. Because the PIFS is shorter than the DIFS (used by the DCF), the PC is considered to have a higher priority to access the medium. Parameters governing the concurrent operation of DCF and PCF, such as the CFP 34

35 CFP Repetition Interval CFP Repetition Interval CFP CP CFP CP PCF DCF PCF DCF CFP Maximum Duration CFP Maximum Duration Figure 2-6: DCF and PCF Superframe Structure Repetition Interval and the CFP Maximum Duration, can be specified to provide a certain QoS. When both DCF and PCF services are desired, durations of Contention Period and Contention-free Period alternate, as illustrated in Figure Basic Frame Exchange The protocol requires that the minimal exchange between two stations consists of two frames. A data frame is sent from the source to the destination, and an acknowledgment (ACK) is returned from the destination to the source, indicating successful receipt of the data frame. Figure 2-7 illustrates a basic frame exchange. This data frame and ACK exchange is an atomic unit of exchange between two stations using the MAC protocol and cannot be interrupted by any other station. To alleviate the problem of hidden nodes in the network, a station also has the option in the basic protocol of using two additional frames, as depicted in Figure 2-8, to notify other stations of the upcoming frame transmission so they delay their own transmissions. The source station sends a request-to-send (RTS) frame, and in response, the destination station sends a clear-to-send (CTS) frame. Upon receipt of the CTS, the source proceeds to send the data frame as above. If the destination correctly receives the frame, it sends an ACK, completing the protocol. This fourframe exchange is also an atomic unit that cannot be interrupted by any other station. 35

36 DIFS Source Backoff Data SIFS Destination ACK Others ;;; ;;; Backoff NAV DIFS Backoff Contention Defer Access Contention Figure 2-7: Basic Frame Exchange DIFS Source Backoff RTS Data SIFS SIFS SIFS Destination CTS ACK Others ;; ;; Backoff NAV (RTS) NAV (CTS) DIFS Backoff Contention Defer Access Contention Figure 2-8: Frame Exchange Using RTS/CTS 36

37 Chapter 3 Theoretical Analysis In this theoretical analysis of the protocol, the efficiency of this protocol in using the wireless medium is determined. Data efficiency is defined as the percentage of total time used for successful transmission of data that the channel is occupied by the actual data. The data efficiency is analyzed for each of the two access mechanisms of From the data efficiencies determined, an upper bound on the throughput is derived. 3.1 Analysis of the DCF Data Efficiency The data efficiency of the DCF mechanism in the protocol can be determined by analyzing the sequence of events that occurs for a basic frame exchange over the wireless medium (illustrated in Figure 2-7). Propagation delay is assumed negligible and is ignored in this analysis. For simplicity, it is also assumed that RTS/CTS is not used. For a successful transmission, the following sequence of events occurs: 1. The medium is idle for a DIFS. 2. The sending station performs backoff. 3. The sending station transmits a packet. 37

38 4. A SIFS interval passes. 5. The receiving station transmits an ACK. The duration of time required for the entire sequence of events can be represented by the sum of the durations of each event. T DCF sequence = T DIFS + T backoff + T packet + T SIFS + T ACK T SIFS and T DIFS are simply the inter-frame space timing interval specified by the protocol for the specific physical layer. For simplicity, a one-stage backoff is assumed where the contention window is always at its initial, minimum size of CW min. Because the backoff value is selected from a uniformly distributed interval from 0 to CW min, the average selected backoff value is CW min 2. The average time required for backoff can thus be calculated as T backoff = average backoff slot time = CW min 2 slot time. The time required for transmission of a packet T packet includes time for transmitting the actual data payload bits as well as all necessary MAC and physical layer overheads. (Please refer to the IEEE data frame format depicted in Figure 2-2 for the fields of a MAC Protocol Data Unit (MPDU).) MAC overheads consist of the MAC header and the FCS. The entire MPDU(MAC header, payload, and FCS) is transmitted at the channel transmission rate. Physical layer overheads include the PLCP-Preamble and the PLCP-Header. Physical Layer overheads are transmitted at the basic rate 1. The transmission time of l bits using a transmission rate of R bits per second (bps) is calculated by l. Thus, the time required to transmit the packet R 1 The basic rate refers to one of the rates in the BSSBasicRateSet. The BSSBasicRateSet is the set of data rates at which all stations in the BSS must be able to receive packets. 38

39 can be expressed as T packet = l MAC header + l payload + l FCS R trans + l PLCP Preamble + l PLCP Header R basic Similarly, the time required for transmission of the ACK can be calculated as the time required to send the ACK frame as well as the physical layer overheads. An ACK frame is transmitted at the basic rate [1, section 9.6] so it can be decoded by all stations. The physical layer overheads are as described for the transmission of the packet above. Thus, T ACK = l ACK R basic + l PLCP Preamble + l PLCP Header R basic To determine average data efficiency, the amount of time spent in the transmission of actual data bits T data must be determined. T data can be calculated by l payload R trans where l payload is the length of the data payload in bits, and R trans is the transmission rate. Using these formulas, average data efficiency is simply average data efficiency = T data T DCF sequence 100% Theoretical Upper Bound of NetworkThroughput To calculate the theoretical upper bound of network throughput, we assume that no collisions occur and all packet transmissions are successful. The maximum throughput is achieved when, immediately upon completion of one sequence of DCF events, the following sequence begins without allowing any idle periods on the wireless medium. Furthermore, for the upper bound of network throughput, T backoff =0. Thismay occur because stations have selected a backoff value of 0 or all packets arriving from higher layers arrive at the beginning of the idle DIFS interval and thus are not required by the standard to backoff. The upper bound of network throughput can then be 39

40 calculated simply as upper bound (throughput) =data efficiency(t backoff =0) transmission rate Calculated Values Table 3.1 lists the values of protocol parameters used in the theoretical analysis. For this analysis, an b DSSS physical layer is assumed. Parameter Value slot time 20 µsec SIFS 10 µsec DIFS 50 µsec CW min 31 PLCP-Preamble 144 bits PLCP-Header 48 bits MAC Header 24 bytes FCS 4bytes ACK Frame 14 bytes Table 3.1: Protocol Parameters for DCF using a DSSS Physical Layer Using these values, the data efficiencies and effective throughput are calculated for different packet sizes transmitted at the various transmission rates supported by b. A basic rate of 1 Mbps is used for these calculations. Table 3.2 shows the calculated data efficiency while transmitting the maximum-sized packet (without encryption) allowed by the IEEE protocol. Transmissions of this packet size produce the highest data efficiency of the protocol. The upper bound on data efficiency assumes a backoff of 0 slots while average data efficiency assumes a backoff of CW min slots. 2 Due to packet overheads, on average, only 94.42% of the bandwidth can be used for transmission of actual data bits. As transmission rates increase, the relative overhead from the physical layer and the ACK packet also increase because these overhead bits must still be transmitted at the slower, basic rate. This results in lower 40

41 data efficiencies at higher transmission rates, with only an average of 65.40% of the bandwidth being used for payload data transmission at an 11 Mbps transmission rate. Transmission Rate [Mbps] Upper Bound on Data Efficiency [%] Upper Bound on Data Throughput [Mbps] Average Data Efficiency [%] Effective Data Throughput [Mbps] Table 3.2: Data Efficiencies and Effective Data Throughput for Sending 2304-byte Packets Using DCF In reality, these values will actually be even lower due to other overheads not included in this analysis such as RTS/CTS packets, Beacon packets, and other control packets such as those used for power management and BSS association, and also due to higher layer protocol overheads such as TCP and IP headers. Furthermore, failed transmissions due to channel conditions will also adversely affect throughput. Fragmentation can help to alleviate the frequent losses of packets from channel errors by minimizing the cost of each loss. With smaller packets, fewer data bits would be lost should the packet be corrupted during transmission. However, the tradeoff of using smaller packets is that the data efficiency is also decreased. Tables 3.3 and 3.4 illustrate some of the effects of packet size on data efficiency and achievable network throughput. Table 3.3 lists the calculated data efficiency for a 1500-byte packet sent at various transmission rates. This packet size represents a maximum-sized IP datagram. Table 3.4 lists the calculated data efficiency and effective network throughput for a 32.5-byte packet. Voice calls using GSM encoding (13 kbps) with 20 msec frames have packets of this size. For both cases, the basic rate is assumed to be 1 Mbps. Due to the high overheads required in the MAC and physical layer for each transmitted packet, transmissions of small packets can be extremely inefficient. For example, a station transmitting 32.5-byte packets over the wireless network at a rate of 11 Mbps would feel as if the network could support a transmission rate of only

42 Transmission Rate [Mbps] Upper Bound on Data Efficiency [%] Upper Bound on Data Throughput [Mbps] Average Data Efficiency [%] Effective Data Throughput [Mbps] Table 3.3: Data Efficiencies and Effective Data Throughput for Sending 1500-byte Packets Using DCF Transmission Rate [Mbps] Upper Bound on Data Efficiency [%] Upper Bound on Data Throughput [Mbps] Average Data Efficiency [%] Effective Data Throughput [Mbps] Table 3.4: Data Efficiencies and Effective Data Throughput for Sending 32.5-byte Packets Using DCF kbps. The smaller the packet, the greater the relative overhead, and thus the lower the data efficiency and data throughput DCF Contention Among Several Stations The above analysis of average data efficiency assumes that each packet being transmitted over the wireless medium will experience an average backoff of CW min slot 2 time intervals before transmission is attempted. While this is true for each station individually, due to multiplexing that occurs among several stations backing-off, the idle periods of backoff seen on the wireless medium during contention between packet transmissions is actually less than the average backoff of CW min. 2 Two stations may each select a backoff value, BK 1 and BK 2, from the uniform distribution between the interval of [0,CW min ]. Suppose BK 1 <BK 2. Station 1 s backoff timer expires first, and it transmits its packet. Station 1 then selects a new backoff value for the following packet and begins backing off once again. Meanwhile, Station 2 had also decremented its backoff counter to (BK 2 BK 1 ) before Station 42

43 1 s transmission. Once the medium is idle again following Station 1 s transmission, Station 2 continues backing off with the same backoff timer. Contention begins once again. However, this time it is between two stations where one has selected the backoff value from the uniform distribution between [0,CW min ], and the second has effectively selected from the uniform distribution between [0,CW min BK 1 ]. Overall, the two stations have equal access to the medium because they both contend using identical methods. Thus, for simplicity, it is assumed that the two stations alternately acquire the wireless medium. In the steady state, the duration of the contention period between packet transmission is the final continuous backoff period of a station before it acquires the medium. This final backoff period (BK) is the minimum of two random variables, one selected from the uniform interval [0,CW min ] and the second selected from the uniform interval [0,CW min BK] where BK is the mean of the final backoff period. The average backoff period between packet transmissions on the medium BK when there are two stations contending is thus BK = 31 BK i=0 i [ ( ) 1 32 ( ) ( ) ( 32 BK i 31 i + 32 BK BK where the expression between the square brackets represents P robability{bk = i}. This expression can be extended for various numbers of stations contending for access to the wireless medium. The MATLAB code used to calculate the average final backoff period between packet transmissions for various numbers of contending stations is included in Appendix A. Figure 3-1 plots BK as it varies with the number of stations contending to access the medium. For one station, BK = CW min =15.5, 2 and it decreases as the number of stations increase. Because the contention period between packet transmissions seen on the medium becomes shorter as the number of stations increase, a higher data throughput can be achieved. The data throughput gradually approaches the theoretical upper bound where the contention period is 0. Figure 3-2 illustrates how the maximum achievable throughput varies for stations transmitting 1500-byte packets at 11 Mbps. 43 ) ]