1

The Impact of MAC Buffer Size on the Throughput Performance of IEEE 802.11
Kamil H. Suleiman, Tara Javidi, Mingyan Liu, and Somsak Kittipiyakul

Abstract The medium access control protocol of IEEE 802.11 networks has been extensively studied to explore its throughput performance. A popular regime of study is the saturation regime where users are assumed to be saturated with information bits. The throughput achieved in this regime, called the saturation throughput, is commonly interpreted as the maximum achievable throughput for any given system. In this paper, we formalize the conditions under which saturation throughput is indeed the maximum achievable throughput. We provide specic settings which may increase the maximum aggregate throughput of the system beyond the saturation throughput. Furthermore, we observe and prove that reducing the MAC buffer size signicantly increases the maximum aggregate throughput, beyond the saturation throughput, a fact that seems counter-intuitive. We formally establish that under small buffer conditions and under UDP-type trafc, a reduction in effective collisions due to the buffer size choice has a positive effect on the throughput performance. This is despite the fact that some packets are dropped because of likely buffer overow events. In other words, by identifying the notion of saturation throughput as an inherently pessimistic notion, this paper brings to the attention the signicant impact of an optimized choice of MAC buffer size on the performance of 802.11 networks. Index Terms IEEE 802.11, medium access control, statistical multiplexing, saturation throughput.

I. I NTRODUCTION With the wide deployment of wireless LAN, its core enabling technology, the IEEE 802.11 medium access control (MAC) protocol, has been extensively studied in recent years. These studParts of the results were presented at the 2005 Allerton Conference K. Suleiman is with the University of Cape Town; T. Javidi and S. Kittipiyakul are with the University of California, San Diego; M. Liu is with the University of Michigan, Ann Arbor.
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ies include the performance as well as fairness properties of 802.11 basic and DCF (distributed coordination functions) options. Many of these studies examine the behavior of a xed number of 802.11 users (or stations/clients) under a special operating regime known as the saturation regime; notable examples include [1]. This is a scenario where all users in the system are innite sources, i.e., they always have packets to send or equivalently, they have innitely many packets waiting in the queues, thus the term saturation. Saturation studies typically characterize the saturation throughput of a system, which is the throughput that each queue or the system as a whole can achieve under the saturation scenario. Saturation throughput is shown to vary with the number of users in the system [1]; they reect in a sense the capacity of the system and provide signicant insights in understanding the limiting behavior of 802.11. Bianchi in [1] rst proposed a Markov chain based modeling scheme to estimate the saturation throughput (these quantities were then used to optimize the backoff window size). Similar models and extensions have since been used for a variety of purposes. Examples include to tune the protocol [2], to study the retry limit [3], or to study different aspects of the saturation throughput, e.g., in the presence of transmission error [11]. The notion of saturation throughput was also studied in [4] as a xed point problem where its existence and uniqueness were examined. More recently, several studies have considered a different system operating regime, where user queues are fed with nite arrival rates. This means that any given user might not always have a packet to send. We refer to this type of queues as nite sources to distinguish them from the innite sources/queues used in the saturation studies. These include studies that propose similar Markov chain based models as originally proposed in [1] while taking into account the fact that the sources may be nite; examples include [5], [6], [12][19]. In particular, in our previous work [5], [7], we investigated how the system behaves when these nite arrival rates approach the saturation rate (throughput) from below, and made the following interesting observation: The queues have very good delay performance even when all queues have arrival rates approaching the saturation throughput1. Building on the above observation, the present paper is an attempt to achieve higher than saturation throughput. This idea builds on our prior work in [5] where we rst proposed practical settings to achieve the saturation throughput. Specically, in the rst part of the paper, we answer
1

This is consistent with other non-saturation studies in literature [6].

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the following question rst posed in [5]: Under the more realistic operation scenario of nonsaturation, is it possible for every user to achieve a throughput higher than that of saturation? Unfortunately, the answer to this question is negative. We show that, in fact, the low delay behavior is due to an extremely long, transient phase in a typical simulation study and is due to the existence of a broken ergodicity phenomenon. Instead of limiting the above observation to a practical lesson on how to set up simulations correctly, we take a further step. The mentioned broken ergodicity phenomenon motivates the main contribution of the second part of the paper: We show that, in the case of realistic trafc (both TCP and UDP sources), reducing buffer sizes of each user signicantly increases the throughput of the system! This increase in overall throughput with the reduction of buffer size might seem counter intuitive at the rst glance, but is explained as follows: The limitation on the buffer space for each user imposes a regular visit to the all-empty queue state and, consequently, to the desirable transient phase with desirable delay behavior. In fact, through a simple Markov chain model, we prove that, even for a buffer size of 1, the total throughput higher than the saturation throughput can be guaranteed for Poisson (UDP-type) source, despite an increase in packet drop probability. One might wonder how the above result depends on the reactive behavior of the arrival stream, the size of the MAC buffer, or the bursty nature of the trafc. The contribution of the present paper is to provide answers to these questions. In the case of TCP trafc, we present some simulation results quantifying the impact of small buffer on the performance of multiple concurrent TCP ows. We will see that even though the packet drops due to MAC buffer limitations result in smaller congestion windows for TCP ows, the consequent lower level of contention achieved by limiting the MAC buffer results in signicantly smaller RTT values, which keep the TCP throughput at roughly similar levels, across various choices of MAC buffer size. This highlights an improved cross-layer interaction via TCPs adaptive mechanism. The rest of the paper is organized as follows. In Section II, we describe our network model and motivate our study using experimental results. Specically, we consider the behavior of a system with symmetric users as the input trafc rate approaches the saturation throughput from below. We shall see that the system has very good delay performance. This prompts the question of whether the notion of saturation throughput is too pessimistic and whether higher than saturation throughput might be obtainable. We attempt to address this question in Section III, and show that for an innite-buffer scenario the answer is negative. We then present a detailed explanation
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802.11 Network

Client Node 1 Client Node 2 802.11 Access Point Wire Destination Node

Client Node 3

Client Node N

Fig. 1.

Network model.

of the good-delay-performance-paradox, and motivate the idea of reducing the MAC buffer size. Then, in Section IV, we theoretically show, using a simple Markov model, how the reduction in the MAC buffer size leads to signicant improvement in throughput (higher than saturation throughput). The same is examined using numerical simulation in Section V for both UDP and TCP trafc sources. We conclude the paper in Section VI. Before we close this section, we would like to acknowledge many important works in redesigning the 802.11 MAC mechanism in general, such as [20], and more specically in improving its throughput performance, e.g., by using rate adaptation [10], by reducing the amount of time spent in backoff [9], and so on. Some of these schemes require modication to the current standard. By contrast the solution presented in our paper does not require any changes, and can be used in conjunction with other solutions. The clean slate redesign of decentralized MAC layer mechanisms with improved performance remains an interesting area of future work. II. N ETWORK M ODEL In this paper, we consider the network scenario illustrated in Figure 1. It consists of a xed number of N wireless nodes that serve as trafc sources, a single access point (base station), and a destination node connected to the base station through a wire (for simplicity). We assume that users have symmetric channels (with negligible error in transmission) and a nite buffer size, ranging from 2 to 10, 000 packets.
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TABLE I IEEE 802.11 PARAMETERS

slot time SIFS DIFS CWmin CWmax retry limit physical layer data rate

20 s 10 s 50 s 31 1023 7 11Mbps

A. Simulation Setup The network simulator (ns2) release 2.28 is used for all the simulation presented here. In a single simulation, the client nodes send theirs packets to the access point, using IEEE 802.11 MAC scheme. Each client node sends packets with an average rate of . The burst of data packets from each node is generated according to the trafc model under study and the size of each packet is xed at 1024 bytes. The aggregate throughput of the wireless LAN was measured by counting all the data bits that arrive at the wired node (or equivalently the base station). In the simulation, RTS/CTS is disabled. This means that the basic distributed coordination function or DCF of 802.11 is used. In all of the scenarios, the buffer size at the base station node was xed at 109 packets. In each simulation, all the client buffers were initialized to empty. The relevant simulation parameters used are listed in Table I. B. Saturation Throughput In Figure 2, we show the dependence of aggregate throughput on the number of client nodes for a buffer size of 10,000 packets which is large enough to be considered innity for our simulation time and topology (the throughput data were collected until all the buffers were nonempty or saturated in our simulations). The saturation throughput or saturation rate is dened as the instantaneous throughput of the overall system when all users at all times have innite number of packets to send. For instance, we note that the saturation throughput for N = 25 client nodes is approximately 4.41 Mbps. Note that, as shown in Figure 2, this quantity is a function of the
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5.3 5.2 5.1 Saturation throughput (bps) 5 4.9 4.8 4.7 4.6 4.5 4.4

x 10

10 15 Number of client nodes

20

25

Fig. 2.

Saturation throughput for different number of client nodes.

total number of sources in the system. Also note that these results are essentially a reproduction of the saturation throughput results reported in [1]. In the next section, we discuss the throughput behavior of the system when the users do not always have packets to send, due to their nite buffer sizes and random trafc arrivals. We call these users nite users or sources. III. T HE F INITE S OURCE PARADOX
AND ITS

E XPLANATION

In this section, we provide an interesting observation: The network outlined above in presence of nite rate sources enjoys very good delay performance even when all sources have arrival rates approaching the saturation throughput. At rst glance, this counter intuitive and rather surprising observation seems to suggest that higher than saturation throughput can be achieved for nite sources. However, we prove this is not the case and provide an intuitive explanation behind this phenomenon. Despite the fact that this is a negative result, our explanation motivates a method for inducing higher than the saturation throughput, which we explore in the subsequent section. Consider the network shown in Figure 1 with symmetric clients and innite buffers. Figure 3 shows the simulated expected number of backlogged clients (out of a total of N = 25 clients), as the arrival rate (load) increases. Here the aggregate arrival rate is normalized by the saturation

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8 7 Average Number of Backlogged Queues 6 5 4 3 2 1 0

0.2

0.4

0.6 0.8 Aggregate Arrival Rate

1.2

1.4

Fig. 3.

Expected number of clients backlogged (out of a total of 25) versus arrival rate normalized to the saturation rate.

throughput of a system of 25 users (roughly 4.41 Mbps). All queues are initially empty and the average number of backlogged queues is computed by collecting data over a simulation run time of 60 minutes. What we see here is that the observed average number of of backlogged queues is much lower than anticipated even when the arrival rate is at or slightly above the saturation throughput. In other words, for a system of N symmetric users, given an empty initial state, even at the aggregate arrival rate of N = sat (N) + , for 0, we do not see, more than eight backlogged queues (less than 1/3 of all queues). Similar results were also reported in [6], where the authors report a large gap between simulated and analytic calculations. This is a rather surprising result in the context of queuing systems operation at heavy loading. One might prematurely interpret Figure 3 to suggest that a throughput higher than saturation is achievable, since the observed average delay is much lower than expected. This, however, is not true. The following result shows that each nite user is limited to its share of saturation throughput despite the above observed good delay behavior. Proposition 1: For any Markovian arrival process, which is independent across queues, the maximum achievable average throughput per user is sat (N)/N. Proof: Obviously, for
N i=1 N i=1

i sat (N), we get

i sat (N), where i is the i > sat (N), the maximum

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throughput of user i. This is because aggregate throughput can never be greater than aggregate arrival rate. Therefore we only need to show that for
January 14, 2008 N i=1

achievable average throughput for each user is sat (N)/N. Consider an N-queue system whose states at time t are given by the queue backlog vector q(t) = [q1 (t), q2 (t), , qN (t)], and the embedded Markov chain where t takes integer values. Consider some time t where all queues are backlogged, i.e., mini qi (t) > 0, the instantaneous total service rate of the system is given by sat (N). Since
N i=1

i > sat (N), i.e., the total arrival rate exceeds the service rate,

at time t there is a strictly positive drift with respect to the state of the system. In other words, there exist some > 0 and M > 0 such that for all i, we have E |qi (t + 1) qi (t)| Dene the stopping time T = min {t : qi (t) M, i} . From the irreducibility of Markov chain, we have P (T < ) > 0. Combining (1)-(3), we conclude that this Markov chain consists of only transient states, i.e. E(min qi (t)) = .
i

min qi (t) > M

i

> .

(1)

(2)

(3)

(4)

This completes the proof. The above result says that no collection of nite sources with symmetric trafc can attain throughput higher than the saturation rate. However, our earlier simulation result suggests that the source could sustain arrival rates higher than the saturation rate without signicant queue buildup for a while (rather long durations even). This presents an apparent contradiction. Understanding this discrepancy turns out to be critical in designing mechanisms that allow us to obtain higher throughput as we detail in the next section. To understand the gap between the theoretical result and the simulation observations in Figure 3, consider again the state of the underlying Markov chain describing the queue state of our N-user system, given by the queue backlog q ZN . Note that in such a network, as some queues become empty, they stop competing for the channel, allowing other users to adaptively choose smaller average back-off windows. This results in a strong negative drift on subspaces

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of form S0 = {q : i s.t. qi = 0}, where qi is the backlog of queue i. When the total arrival rate > sat (N), this negative drift creates a broken ergodicity phenomenon: although the system consists of all transient states, the time averages computed over a simulation run time are signicantly different from the ensemble averages. More precisely, there are effectively two classes of states, those with negative drift, S (S0 S ), and those with positive one, S + . If the arrival rate is greater than sat , all states are transient; however, when starting with all queues empty (in S ), the time it takes to rst hit a positive-drift state can be signicantly longer than the simulation time. Indeed, we have never observed such a transition in any of our simulation, which typically runs on the order of hours in simulated time. In other words, for realistic simulation times the ensemble average one wishes to obtain can be drastically different from the computed time average that one ends up getting. To compute the ensemble average, a randomization of initial queue state is required. However, from (4), it has been shown that the mean queue length in each buffer is innity. Another alternative to computing the ensemble average is to allocate a large initial queue length. One might argue that the above phenomenon and explanation are lessons in running credible simulations and reading the results (after all, one needs to understand the underlying random processes before replacing ensemble averages by time averages). But in addition, these results do lead us to think differently about our system. One can essentially view the system as a globally unstable one with a desirable (high throughput) operating point which is, given a particular set of initial conditions, practically robust (i.e., it can stay around the high throughput region for a long time). This robustness depends on the statistics of the stopping time associated with the rst transition from S to S + , which itself heavily depends on the statistics of the arrival processes (beyond the average mean) as well as the transmission and back-off policies, and nally, is determined by the maximum of packets each user holds. This leads to the question of whether one might be able to introduce mechanisms that stabilize the system around the robust equilibrium, while maintaining high throughput, by preventing the system from transitioning to S + . An intuitive way to force the system to stay in S is simply to periodically (but not simultaneously) empty the queues. This would result in a time share among desirable (higher throughput) operating points. In practice, such a behavior may be induced by articially reducing the buffer size at the MAC layer. In the next section, we will rst formalize this idea. In particular, we will use a primitive model for the MAC interaction and
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queue build-ups to hypothesize about a mechanism that enables higher than saturation throughput. In Section V, we will test our hypothesis via extensive simulation. IV. T HROUGHPUT I MPROVEMENT
VIA

R EDUCED MAC BUFFER S IZE

In this section, we consider a symmetric network where each user has a perfect channel, a nite arrival rate , and a nite buffer space. In particular, we will consider a special case where each users buffer capacity is limited to one packet at a time, and show an increase in the achievable throughput. This result suggests that even though we incur a loss when dropping packets due to buffer overow, we are able to compensate this loss with an increase in the departure rate2 ! To facilitate the analysis, consider the following simplied model of the network shown in Figure 4. This is a continuous-time extension of the models proposed in [4]: there are N identical nodes with buffers of size B. Arrival rate to each node is Poisson with rate (if the buffer at a node is full at the arrival time of a new packet, the packet is dropped). Furthermore, the arrivals to nodes are assumed to be independent among nodes. Here we ignore the back-off times and focus only on the times when successful attempts are made, i.e., we assume that when there are k backlogged users a successful attempt leads to the channel being allocated to one of the k contending nodes with equal probability. We also assume that the time between two successful attempts, given k contending nodes, is an exponential random variable with rate sat (k). In this paper, we assume that sat () is a strict monotonically decreasing function, i.e., sat (i) > sat (j) for any 1 i < j. Let () denote the average departure rate (average throughput). Note that from previous sections we know that if B = , then the maximum achievable average throughput is sat (N). In what comes next, we calculate () for the general B 1 case. A straight-forward, albeit computationally-intensive, approach is to use the the number of packets in each queue as the state vector. This approach requires (B + 1)N states and is given as follows: we represent the
2

Note that in practice this loss may not be real, in that packets are typically stored in buffers of higher layers, e.g., transport

layer TCP/UDP buffers, before they are moved to and accepted by the MAC buffer. Therefore, a very small MAC buffer could simply mean that packets may only be transferred into the MAC buffer in very small quantities at a time, e.g., resulting in a small TCP window size as we shall see in the next section.

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11

Randomly select 1 out of nt backlogged queues at rate sat(nt)

nt = number of backlogged queues at time t

Fig. 4.

A simple queuing model to approximate 802.11 MAC. In this illustration, nt = 4.

system as a time-homogeneous continuous-time Markov chain [8] with state vectors s = (s1 , s2 , . . . , sN ) S := {0, 1, . . . , B}N . Here sn is the number of packets in queue n {1, . . . , N}. The time-homogeneous transition rate quv from state u S to state v S is given as follows: , if v = u + en for some n {1, . . . , N} (|u|) sat |u| , if v = u en for some n {1, . . . , N} quv = q , if v = u sS:s=u us 0, otherwise, 1, if A N where we dene |u| := m=1 1{um =0} , indicator 1A = , and en denotes a vector 0, if not A whose N elements are all zero except one in the nth position. We let a matrix Q = [quv ] to denote the innitesimal generator matrix. Since the above continuous-time Markov chain is irreducible and has a nite number of states, there exists a unique steady-state probability vector = [s ]sS such that 0=
uS

u quv , v S,

or in vector-matrix form, 0 = Q.
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Therefore, the average throughput is given as

N N

() =
sS

sat (|s|)s =
m=1

sat (m)
sS:|s|=m

s =
m=1

sat (m)m (),

where m () :=
sS:|s|=m

s is the steady-state probability when there are m backlogged users

in the system. Due to the large state space, the above calculations have a large computation complexity. On the other hand, it is not difcult to calculate for the special case of B = 1. When B = 1, the number of backlogged users, m(t), also forms a continuous-time Markov chain. This allows a closed form characterization of (). The following theorem shows that () can be strictly greater than sat (N) when B = 1. Theorem 1: Assume B = 1. There exists a rate of arrival, such that for all the achievable average throughput () is strictly greater than sat (N). Proof: The number of backlogged users in the system forms a birth-death continuous-time Markov chain with state m {0, 1, . . . , N}. The state transition diagram for such a system is shown in Figure 5. Note that the total departure rate in this model is
N

() =
m=1

sat (m)m (),

(5)

where m () is the stationary distribution for state m, given the arrival rate . On the other hand, from the balance equations, we obtain the following: m () = N () sat (k) , m = 0, 1, . . . , N 1. (N m)!(N m)
N m=0 N k=m+1

(6)

This together with the fact that and m , m = 0, . . . , N 1:

m () = 1 leads to the following expressions for N

1

N 1

N () = = m () =

1+
j=0

N k=j+1 sat (k) (N j)!(N j)

N!N N!N + N!N +

N 1 j j=0 [N]j N s=m+1 N k=j+1 sat (k)

(7) , (8)

[N]m m

sat (s)
N k=j+1 sat (k)

N 1 j j=0 [N]j

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N 0 sat(1) 1

(N-1) (N-m+1) m sat(2) sat(m)

(N-m)

sat(m+1) sat(N)

Fig. 5.

Continuous-time Markov chain describing the number of backlogged users.

where [N]m denotes

N! . (N m)!

Inserting (7) and (8) into (5) gives the achievable average throughput: () =
N m=1

sat (m)[N]m m
N 1 j j=0 [N]j N s=N +1

N!N +

N s=m+1 sat (s) , N sat (k) k=j+1

(9)

where we use the convention that have () = sat (N)

sat (s) 1. Note that, by rearranging the terms, we sat (k)

N 1 k=m+1 sat (k)

N!N + N!N +

N 1 N 1 m m=1 [N]m sat (m) k=m+1 N 1 N m m=1 [N]m sat (N) k=1 sat (k) +

(10)

It is easy to see that () sat (N) as . In addition, because of the monotonic decreasing property of sat (), we note that the numerator in (10) is greater than the denominator, for sufciently large . That is, there exists for which () > sat (N), for any > . Figure 6 compares the above calculation with the simulation result for N = 25 users with B = 1. Even though, as expected, the simple MAC model proposed earlier fails to capture the precise departure rate, our simulations conrm the general trend specied by Theorem 1. In other words, limiting the buffer size to 1 unit improves the maximum achievable throughput of the system. Note that such a buffer constraint, on the other hand, causes packet drops in the admissible regime (when N < sat (N)) which are nonetheless compensated by the increase in the overall throughput rate. It clearly conrms the result of Theorem 1, suggesting that is 5.2 Mbps from calculation and 4.4 Mbps from simulation. This discrepancy between the two graphs is attributed to the overly simplied model of 802.11 MAC. However, the above simple analysis allows us to hypothesize the following interesting and rather counter-intuitive phenomenon: in
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5.5 5 Aggregate throughput () (Mbps) 4.5 4 3.5 3 2.5 2 1.5 1 0.5 0 0 1 2 3 4 5 6 7 Aggregate arrival rate N (Mbps) simulation calculation (N)
sat

Fig. 6. Dependence of aggregate throughput on aggregate sending rate for a system of N = 25 client-nodes. Each client nodes MAC buffer is reduced to one packet. The horizontal dotted line shows the saturation throughput sat (N ).

spite of the introduction of buffer overow and packet loss, the overall throughput of the system increases as the MAC buffer size is reduced. In the next section, we provide numerical and simulations studies, conrming this hypothesis. V. S IMULATION R ESULTS
AND

D ISCUSSIONS

In this section, we demonstrate the validity of our small buffer proposal via simulating a network of 25 users/nodes with nite arrival trafc rates. Parameters are the same as those given in Section II, and the specic values of sat () are obtained from Figure 2. We distinguish between sources of trafc that are reactive (e.g. TCP) and those that are not (e.g. UDP). In subsequent subsections, we rst consider the case of UDP trafc and then the case of TCP trafc. A. Sending UDP Packets Here, we study the dependency of the system throughput on buffer size when the client nodes send UDP packets via the base station. At the MAC layer, we allocate different buffer sizes to client nodes to see the dependency of the aggregate throughput on the buffer size of each node. Figure 7 shows a graph of aggregate throughput versus arrival rate when each client node is allocated with buffer size 1 and 10000, respectively. The arrival process to each node is Poisson.
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x 10

1pkt 10000 pkts 5 Aggregate throughput (bps)

3 4 5 6 Aggregate arrival rate (bps)

8 x 10

9
6

Fig. 7.

Throughput performance for buffer size 1 and 10000 packets, respectively, with Poisson arrivals.

Here, we notice that when the buffer size is 10000, the aggregate throughput is basically the same as the aggregate arrival rate until it peaks at saturation (around 4.6 Mbps). It then follows a similar at curve that is consistent with the saturation curve. In the case of 1-packet buffer size, we notice that the curve overlaps with that of 1000-packet buffer size until the arrival rate exceeds the saturation throughput. At this point, unlike in the previous case, the throughput continues to increase past the saturation point at 4.6 Mbps and starts to drop around 5.2 Mbps (close to a 10% increase in throughput) and gradually returns to the saturation throughput when the arrival rate exceeds 8.5 Mbps. Here we see a signicant increase in throughput when we compare the maxima of the two curves. In particular, the range between 4.6 Mbps (the saturation point) and 8.5 Mbps is the ideal range in which the system should be operated; it is in this range we achieve higher than saturation throughput. Similar results can be obtained for various choices of buffer sizes. Figure 8 shows a zoomedin plot of the system throughput versus arrival rates around the saturation rate. The conclusion remains the same: for arrival rates less than the saturation rate we get essentially the same values of system throughput; for arrival rates higher than the saturation rate smaller buffer sizes result in higher system throughput than the saturation rate. Table II lists the maxima of the throughput curve for the different buffer sizes considered

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5.2 5.1 Aggregate throughput (Mbps) 5 4.9 4.8 4.7 4.6 4.5 4.4 4.5 1 pkt 4 pkts 9 pkts) 30 pkts 10,000 pkts

5.5

6 6.5 7 7.5 Aggregate arrival rate (Mbps)

8.5

Fig. 8.

Throughput performance for different buffer size values TABLE II

C OMPARISON OF THE MAXIMUM THROUGHPUT ACHIEVED FOR DIFFERENT BUFFER SIZES AND
IN THROUGHPUT OVER THE SATURATION THROUGHPUT

THE RESPECTIVE INCREASE

Buffer size 1 4 6 9 30 100 10000

Approximated maximum 5064913 4959693 4877423 4841981 4497785 4497785 4497785

Percentage increase in throughput 14.56 12.19 10.31 9.50 1.71 1.71 1.71

in the above simulation. Note that these maxima are approximate values as the data points are discrete. We also experimented with arrival processes that are more bursty than Poisson. In particular, we also used arrivals that occur in bursts of average size 20 packets, with an exponential interburst time. The results are shown in Figure 9, with a similar observation and conclusion. Note

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5 4.5 4 Aggregate throughput (Mbps) 3.5 3 2.5 2 1.5 1 0.5 0 0 1 2 3 4 5 Aggregate arrival rate (Mbps) 20 pkt 10,000 pkts 6 7

Fig. 9. Throughput performance for buffer size 20 and 10,000 in which packet arrival process is exponential and bursty (average burst size of 20 packets)

that in bursty scenarios, the MAC buffer cannot be reduced beyond the average burst size without the probability of packet drop becoming prohibitively large. From the above simulation results and comparisons, we conclude that saturation throughput is a pessimistic criteria when working with 802.11 network. We also observe that a large buffer size in client nodes can have a negative effect on throughput performance. The intuitive reason is that for arrival rates beyond saturation, the higher the arrival rate, the higher the chances of collision between competing packets. This, in turn, leads to large average window size using exponential back off, which results in waste of bandwidth. Reducing the buffer sizes at the client nodes effectively reduces arrival rates seen by the base station for arrival rates beyond saturation. There are two important points to make in applying the above observation to practice. The rst is that in order to fully benet from the above result, ideally one would like to operate the system around the region where the aggregate throughput exceeds the saturation throughput, by allocating the right buffer size. This may not be easy in practice because one does not necessarily know a priori the number of clients in the system or their respective arrival processes. Fortunately, even if one cannot perfectly t the buffer size to the trafc scenario, not much is lost in term

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of throughput. As we have seen, a smaller buffer gives identical throughput performance as a large buffer under all arrival rates except for the immediate region exceeding the saturation rate, in which case smaller buffers outperform large buffers. Even more fortunate, in the case of transporting using TCP, as we shall see in the next subsection, TCP actually adapts to the smaller buffer size (adjusting its congestion window size) and learns the ideal operating point. The second point is that the gain in throughput for UDP trafc is obtained at the expense of higher packet losses due to buffer overow. We note that, for all buffer sizes and at arrival rates above the saturation rate, the higher the arrival rate, the higher the difference between arrival and service rates. In the case of smaller buffers, this difference means packets are lost due to more buffer overow; whereas in the case of large buffers, this difference means packets are queued indenitely in the buffer (up to the buffer capacity), as well as more loss due to collision. This distinction is important as it sheds light on the advantage of losing packets at the MAC buffer: buffer overow prevents excessive trafc and packet collision over the air. Therefore, reliability mechanism (e.g., packet loss recovery/retransmission) at the higher layers does not involve the actual retransmission of packets over the air. Such a system is much more bandwidth and energy efcient than retransmitting packets lost due to collision. Having demonstrated the benets of adopting smaller buffer sizes at the MAC layer for UDP trafc, we next examine whether the same approach has negative impact on TCP trafc, which is reactive unlike UDP. We will see that the adaptive nature of TCP makes its performance (to a large extent) impartial to the choice of the MAC buffer size. B. Sending TCP Packets The sending rate of TCP packets is determined by the TCP agents at both the sender and receiver. If TCP packets are being sent by client nodes, the sender-side TCP considers any loss of packets as a sign of network congestion and, consequently, throttles its sending rate. The reactive response of TCP, i.e. additive-increase, multiplicative-decrease of the congestion window, adaptively controls the rate at which packets are being lost for any reason. In our network, two possible reasons signicantly contribute to the loss of a packet: buffer overow and timeout (both are regulated and impacted by users interactions and dynamics.). Table III shows the aggregate throughput, number of packets dropped, the average round trip time (RTT) for TCP packets, and the average congestion window during the simulation for
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TABLE III PARAMETERS MEASURED FOR TCP SIMULATION FOR DIFFERENT BUFFER SIZE VALUES

Buffer size 1 3 6 10 30 10000

Throughput 3276458 3280281 3266901 3276458 3275366 3275366

Buffer overow drop 106862 18205 172 12 0 0

Drop due to timeout 2 5 48 25 15 15

Average RTT (s) 0.261492 0.712183 1.21481 1.21821 1.21982 1.21982

Average CWND 3.53879 14.2779 202.856 223.326 224.9 224.9

different choices of MAC buffer sizes. Many important observations and conclusions are to be made here: First, we do not see any signicant differences between the net aggregate throughput achieved across different scenarios. Moreover, while the number of packets dropped due to timeout are quite insignicant, we see signicant and consistent changes in number of packets dropped due to buffer overow, average RTT, and average TCP congestion window size. This can be explained as follows. Packet losses due to timeout do not have signicant impact on the throughput performance in the system. As the buffer size increases, the probability of dropping packets decreases signicantly. This has a positive impact on the TCP congestion window size. However, we do not see any proportional increases in aggregate throughput as the average TCP congestion window increases. This is because, in the case of higher buffer size, there is a higher competition to share the bandwidth which increases the queue delay of packets (shown by an increase in RTT). This slows down the sending rate for a given congestion window as TCP uses a sliding window mechanism. That is, although a packet with such a long RTT is not dropped, it blocks the sliding TCP window. This is exactly why the aggregate throughput does not vary signicantly with buffer sizes. Here we note that the lower aggregate throughput in TCP trafc, relative to UDP trafc, is somewhat expected due to the less stochastic nature of the TCP trafc streams and the lower statistical multiplexing of the sources. This lower statistical multiplexing results in a higher level of simultaneous use of channel and, hence, higher collision rates. In other words, the adaptive nature of TCP streams is both the curse and the blessing: it is a curse in smoothly adapting trafc,
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resulting in higher collision rate and, hence, less efcient use of bandwidth; it is a blessing as it guarantees a robust performance against variations in MAC buffer sizes. The second issue at hand is the impact of packet loss due to buffer overow. We note two regimes for the operation of the network in relation to packet loss. Our studies show that, when trafc intensity is signicantly below saturation, packet loss due to overow is negligible. With an increase in trafc load, packet overow loss increases. When the trafc load is strictly below saturation, the short buffers introduce an undesirable loss phenomenon. This loss probability can easily be combatted at high layers via FEC or packet retransmission3. In the regime where arrival rate surpasses the saturation throughput of the system, the packet loss due to overow becomes insignicant compared to the packet loss due to collisions and deadline expiry. This means that the negative impact of a short buffer, i.e., higher buffer overow rate, is dominated by the positive impact of short buffer on decreasing the collision rate. In other words, this highlights a similar point we made in the case of UDP trafc: that is, for all buffer sizes, the higher the arrival rate (beyond saturation), the higher the difference between the arrival rate and service rate. As mentioned earlier, in the case of small buffers, this difference means packets queued increase buffer overow, whereas in the case of large buffers, the difference means packets queued indenitely in the buffer (up to buffer capacity ) as well as increased packet loss due to collision. This distinction is important because the former does not introduce excessive trafc and packet collision over the air. Therefore, any reliability (e.g., packet loss recovery and retransmission) measures at higher layers do not involve the actual transmission of packets they are lost prior to transmission which is thus much more bandwidth and energy efcient. VI. C ONCLUSION Most studies regarding 802.11 technology have focused on what is known as saturation analysis. In this paper, we rst showed that the notion of saturation is an inherently pessimistic one as it does not fully convey the capacity of the system. Our work was motivated by our observations and simulations of the 802.11 network. We used basic concepts from Markov chain theory to explain a seemingly paradoxical gap between theoretical notion of throughput and
3

The FEC technique is advantageous to retransmission as it does not incur an out-of-order delivery problem, but both techniques

are simple and easy to implement.

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the simulated performance of an 802.11 network. We then relied on this insight to show that a reduction of MAC buffer size, in fact, signicantly improves the throughput performance in the network. The main reason behind this improvement is the monotonically decreasing property of the saturation throughput in 802.11 network with the number of backlogged users. While we have shown that saturation throughput does not completely capture the capability of 802.11, it remains one of the most succinct and natural notions of performance. It would be highly desirable if we could come up with a similarly succinct and natural notion in the non-saturated regime (conceivably the nite buffer size will have to be counted for somehow). This is a potentially interesting future research direction. R EFERENCES
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[14] K. Sakakibara, S. Chikada, and J. Yamakita, Analysis of unsaturation throughput of IEEE 802.11 DCF, IEEE International Conference on Personal Wireless Communications (ICPWC), pp. 134138, 2005. [15] G.-S. Ahn, A. T. Campbell, A. Veres, and L.-H. Sun, Supporting service differentiation for real-time and best-effort trafc in stateless wireless ad hoc networks (SWAN), IEEE Trans. Mobile Computing, vol. 1, pp. 192207, 2002. [16] M. Ergen and P. Varaiya, Throughput analysis and admission control in IEEE 802.11a, Mobile Networks and Applications, vol. 10, no. 5, pp. 705716, Oct. 2005. [17] A. Zaki and M. El-Hadidi, Throughput analysis of IEEE 802.11 DCF under nite load trafc, in Proc. First Inter. Symp. on Control, Commun., and Signal Processing, pp. 535538, 2004. [18] O. Tickoo and B. Sikdar, A queueing model for nite load IEEE 802.11 random access, in Proc. IEEE Inter. Conf. on Communications, vol. 1, pp. 175179, June 2004. [19] K. Duffy, D. Malone, and D. J. Leith, Modeling the 802.11 Distributed Coordination Function in non-saturated conditions, in IEEE Communications Letters, Vol. 9, No. 8, August 2005. [20] P. Gupta, Y. Sankarasubramaniam, and A. Stolyar, Random-access scheduling with service differentiation in wireless networks, in Proceedings of the 24th Inter. Conf. on Computer Communications (INFOCOM), Mar. 2005.

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