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809

A New Packet Loss Model of the IEEE 802.11g Wireless Network for Multimedia Communications

Luis Carvalho, Joao Angeja, and Antonio Navarro, Member, IEEE

as inaccurate in [16] and [17]. In [16], a theoretical approach and computer simulations were conducted to show the need to extend the number of Markov states to more than two states in order to characterize a Rayleigh fading channel adequately. The authors in [17] claim that the GE model is a bad approximation to IEEE 802.11b [18] wireless networks due to the existence of heavy-tailed run (gap) and burst length distributions. The same phenomenon was observed in our experiments as far as IEEE 802.11g is concerned. They showed the advantages of the Chaotic map and the semiMarkov models. However, these models are more complex than the proposed one which has similar complexity as GE. Furthermore, the proposed model surpasses GE even in lighttail situations. In this paper, we propose a novel model for IEEE 802.11g packet wireless networks with more accuracy in terms of fitting the heavy tail, and with similar complexity as GE. This paper is divided as follows. Section II reviews the GE model. In Section III, we describe our model, and in Section IV, we present some experimental results. In Section V, we prove that our model is superior to GE. Section VI shows the dependency between the proposed model parameter and packet rate under low and medium bit rates usually used in compressed video streaming. Finally, Section VII draws some conclusions. II. GE MODEL In packet transmission over many real networks, the errors occur in bursts separated by error-free gaps. In a communication network with no-guarantee of quality of service, characteristics like the channel deep fading and memory overflow are the main causes of such bursts of lost and erroneous packets. Despite communication systems may use interleaving to smear out the errors, the channel decoders may introduce bursts of error when the number of errors is greater than the decoder correction capability. A model of the memory effect resulting from bursts of errors was suggested by Gilbert [1] and Elliott [2]. It is a two-state Markov model with one good (G) state and one bad (B) state as shown in Fig. 1. Each state corresponds to a specific channel quality and behaves as a binary symmetric channel (BSC) with a certain crossover probability Pg for the G state and Pb for the B state. Therefore, Pb>Pg.

Abstract A wireless indoor network model coping with packet loss and packet arrival is proposed. This novel model is compared to the well known Gilbert-Elliott (GE) model. The proposed model is shown to be more accurate than GE in a particular but very representative IEEE 802.11g scenario. The proposed model has two parameters: one is a function of the source packet rate; and the other is almost constant. The extensive usage of the GE model in data communications is due to its simplicity. Likewise, it is expected a similar adoption of our model for wireless multimedia packet communications, since it was conceived based on traffic with multimedia features1. Index Terms Multimedia packet communications, wireless LAN modeling, IEEE 802.11g.

I. INTRODUCTION Communication system modeling allows researching novel solutions to increase channel capacity. Besides, in the context of network convergence and interoperability, communication system modeling and simulation allow us to evaluate, in a fast manner, the end-to-end communication performance as well as optimally tuning terminal and network parameters. Thus, in this context, the network model accuracy is crucial. Any network is usually characterized statistically by several parameters. This paper proposes a statistical model for the number of consecutive arrived and lost packets as simple as the famous GE mathematical model [1], [2], but more accurate, at least for the wireless indoor network IEEE 802.11g [3]. The GE model has extensively been used to assist in the design of communication systems. In [4]-[6], channel codes were evaluated in the presence of the GE channel. Frossard and Verscheure [7], and Stuhlmuller et al [8] proposed a solution for the joint video source-GE channel coding problem for MPEG-2 [9] and H.263 [10], respectively. Gnavi et al [11] optimally designed some coding parameters for H.264 algorithm [12]. It was employed a GE model to drive the selection of those coding parameters. More recently, in the case of image transmission, Grangetto et al [13] also solved a code rate allocation problem and demonstrated a solution employing SPIHT [14] and JPEG2000 [15] as source codecs and GE as a wireline Internet network model. Despite its high adoption in the study of communication systems at link, transport and application layers, the GE model was pointed out

1 The authors are with the Telecommunications Institute, University of Aveiro, 3810-193 Aveiro, Portugal (e-mail: navarro@av.it.pt).

810

1 G 1

Fig. 1. GE model.

P[X = n] = p n =

B

k n , n = 1,2,3,... n

(6)

At each packet interval, the channel changes to a new state with transition probabilities of going from state G to state B equal to 1- and from state B to state G equal to 1-. In our comparative study, we assumed Pg=0 and Pb=1 [4]. Let random variables X and Y be the error burst length and the error-free (gap) length, respectively. Thus, from the GE model, the probability mass functions (pmf) of X and Y are geometric and given by [19],

k=

1 . ln(1 )

(7)

The mean, E[X], and variance, V[X], of (6) are, respectively, given by,

E[X] =

k (1 )

.

(8)

P[X = n ] = n 1 (1 ) P[Y = k ] = k 1 (1 )

(1) (2)

V[X] =

k(1 k) (1 ) 2

(9)

E[X] = 1 /(1 ) (3) (4)

As we need to estimate from the observed data, we determine the maximum likelihood estimation (MLE) of , . The MLE is given by maximizing the following i.e., expression,

E[Y ] = 1 /(1 ) .

From (3) and (4), the packet loss (PL) is expressed by,

L=

p

i =1

n i ;

(10)

(5)

where ni, i=1,2, is a set of observations. Substituting (6) into (10), we get

which is equal to the probability of being in state B in a stationary situation. From some experiments, we could compare (1) and (2) to our proposed statistical models. In Section IV, we will show that there is no significant crosscorrelation between X and Y.

L=

i =1

n i k N i=1 k = N ni ni

n i

(11)

i =1

Keeping in mind that the transport layer will encapsulate real time multimedia, we propose modeling the length of consecutive erroneous or correct (arrived) RTP [20] packets transmitted over UDP/IP/802.11g protocol stack. To achieve this goal, we performed the chi-square goodness-of-fit test of a geometric (GE model) and a logarithmic series distribution (proposed model) to a set of experimental data. The logarithmic series distribution was chosen after testing several known distributions, which did not fit well the experimental data. Let X be the burst length of lost RTP packets and suppose that X is a logarithmic series random variable with pmf expressed as,

= ln(L) = N ln(k) +

i =1

) - ln( n i ln(

n ) .

i i =1

(12)

For simplicity we replace the three right terms of last expression by f1, f2 and f3, i.e.,

(13)

(14)

L. Carvalho et al.: A New Packet Loss Model of the IEEE 802.11g Wireless Network for Multimedia Communications

811

where,

) 1 f 2 (n, =

n

i =1

(16)

f 3 (n) =0.

By setting (14) equal zero,

(17)

Nk 1 = + (1 )

n

i =1

=0

(18)

Initially, we selected three RTP payload sizes. Two of them, payload sizes of 50 and 200 bytes are related to two common audio compressed bit rates, 4 kbps-10 packets/s (R1) and 13.3 kbps-8.33 packets/s (R2), respectively. The third, an 160 bytes payload size is related to video coding at 256 kbps-200 packets/s (R3). Nevertheless, for the time being, as we are not analyzing at the application layer, all RTP payloads are filled with random data. In Section VI, we will also consider other packet rates. We set up seven scenarios, shown in Table I, with two stations, Sta1 and Sta2, intercommunicating via an access point (AP). In scenarios S1x, only a very low bit rate (audio) is exchanged between stations. Both stations deliver at either R1 (S11) or R2 (S12). In scenario S21, both stations deliver only video at R3. In scenario S31, Sta1 delivers audio to Sta2 at R1 and receives video from Sta2 at R3. In scenarios S4x, both stations deliver audio plus video to each other. Particularly, in S41, Sta1 delivers audio and video at R1 and R3, respectively. In all scenarios, the packets are retrieved at a constant packet rate.

TABLE I TESTED SCENARIOS. DELIVERED BIT RATE. Scenarios Station 1 Station 2 S11 R1 R1 S12 R2 R2 S21 R3 R3 S31 R1 R3 S32 R2 R3 S41 R1+R3 R1+R3 S42 R2+R3 R2+R3

1 N

k , ni = ) (1 i =1

N

(19)

where N is the total number of observations. Observing (19), we realized that the MLE of is determined using the expectation expression (8). Our final model is described by two logarithmic series pmfs, one for lost packet burst and another for the number of consecutive arrived packets denoted by P[X] and P[Y], respectively. Equation (19) will then be used to find an estimative of from observed runs and bursts. As shown, in Section IV, from all collected data, X and Y are independent random variables.

After carrying out several field measurements, we analyzed statistically the burst length of packets loss (X) and run length of packet arrival (Y), separately. We estimated the distribution by averaging all observations and substituting (7) parameter into (19). As far as we know there is no reference testbed for 802.11g published in the literature. However, we tried to set up one very representative based on our laboratory dimensions.

All scenarios were implemented on an infrastructure mode [3] as depicted in Fig. 2 where station 1 (Sta1), via AP, accesses media data from station 2 (Sta2) and vice versa. Significant burst lengths are achieved only if the network traffic is above 10 Mbps. Therefore, in all scenarios, Sta3 and Sta4 are exchanging data between them, through the wireless network infrastructure, each sending a source bit rate of 6.1 Mbps (RTP payload). These two stations load the network increasing the source traffic and thus acting like simulators of tens of other interfering stations.

10 meters

A. Experimental Testbed In this sub-section we describe the basic scenarios used in our experiments. For these measurements, we considered sending RTP/UDP/IP [11] packets over 802.11g WLAN since we are interested in the transportation of real time compressed multimedia signals. We have not used any specific standardized RTP payload. Instead, we were only interested in taking into account all packets overheads. Furthermore, any erroneous packet is discarded at UDP/IP level. The packet loss was monitored by checking the sequence number field in the header of the received RTP packets.

AP

Sta 3

Sta 4

, for all scenarios Table II shows the estimated parameter, under evaluation. The random variable X denotes the lost packet burst length (number of consecutive lost packets) and Y the received packet gap length. The bit rate and packet rate values used in each scenario are presented in Table I and are

Wall

812

0.25 Observed Data CAN model GE model 0.2

related to the RTP payload sizes. Each table cell under either values for each scenario corresponding to X or Y shows either the error burst length or the error-free gap (run) length, respectively. As already referred, for instance, in S11 audio data at 4 kbps (R1) is exchanged between both stations, Sta1 equal and Sta2, resulting in a burst length distributions with to 0.292 and 0.287 measured at Sta2 and Sta1, respectively. As another example to explain Table II, in S41, Sta1 streams at = 0.773 and R1 and R3 to Sta2, corresponding to

was estimated from = 0.902 , respectively. In Table II, (19). We should note that all measurements were actually carried out at the receiving station.

TABLE II PARAMETER Scenarios S11 S12 S21 S31 S32 S41 S42

10 k

15

20

25

Station 1 X Y

0.292 0.278 0.937 0.463 0.387 0.773 0.902 0.735 0.902 0.992 0.982 0.991 0.925 0.929 0.737 0.847 0.734 0.838

Station 2 X Y

0.287 0.314 0.937 0.784 0.738 0.754 0.868 0.735 0.902 0.992 0.981 0.992 0.958 0.975 0.709 0.857 0.734 0.838

We calculated the crosscorrelation function (i-j) of the observed data, Xi and Yj, i, j {0,1,2,,2750}, showing very low correlation, |(i-j)|<0.06 as shown in Fig. 5. Likewise, the autocorrelation functions of Xi and Xi+m, and of Yi and Yi+m, m 0, give no significant values, |(m)|<0.05. Therefore, Xi and Yj are two independent identical distributed random variables denoting two wide-sense stationary discrete-time random processes.

0.08

(i j)

0.06

values are close to 1 Observing Table II, we realize that for Y and close to 0.3 for X under low bit rate source traffic. It means long runs and short bursts. However, as the traffic . For instance, increases the burst lengths increase as well as

in scenarios S3x, Sta2 delivers at rate R3 and receives at lower values for X are around rate, either R1 or R2 and therefore,

values for Y are always above 0.75 and 0.4, respectively. 0.7 thereby long runs or heavy-tailed behavior.

0.35 Observed Data CAN model GE model 0.3

500

(i j)

0.25

V. CHI-SQUARE TEST

The natural test to carry out is an eyeball comparison of the postulated pmf and the experimental determined counterpart (see Fig. 3 and 4). However, we should follow a more formal way of comparison. We have performed a chisquare test for each scenario described in Table I. The chisquare statistics [21] is defined as the weighted difference between the observed number of outcomes, Nn, n {1,2,...,I} and the corresponding expected number Mn,

P[X=n]

0.2

0.15

0.1

0.05

10 n

15

20

25

B. Experimental Results To illustrate graphically how good our model fits to collected data, Fig. 3 and 4 show both models and observed data related to X and Y, respectively, in scenario S21 (Sta2). Equations (3) and (4) were used to estimate the GE model parameters, and .

Z=

(N n M n ) 2 Mn n =1

(20)

L. Carvalho et al.: A New Packet Loss Model of the IEEE 802.11g Wireless Network for Multimedia Communications

813

where,

M n = pn N

(21)

and I is the number of intervals. For large N, the random variable Z has a chi-square pdf with I-2 degrees of freedom.

TABLE III CHI-SQUARE TEST RESULTS Station 1 Station 2 Scenarios X Y X Y Prop GE Prop GE Prop GE Prop GE 0.24 2.96 63.87 452.78 0.58 3.66 68.06 532.83 S11 1.98 9.92 158.68 190.70 0.74 1.37 107.44 253.67 S12 1074.71 2074.45 335.24 10231.49 1657.75 1368.62 306.66 8977.31 S21 19.84 5.28 132.70 101.54 1803.06 94.75 670.79 6753.68 S31 4.51 20.64 344.69 64.76 1409.05 117.90 947.17 3579.09 S32 S41 S42

50.01 1635.75 55.35 1485.97 100.28 2007.84 19.44 2109.97 101.74 278.02 63.03 208.03 3.92 2822.73 31.83 6374.76 60.99 2495.56 55.35 1485.97 48.28 7354.64 19.44 2109.97 94.04 707.65 63.03 208.03 3.29 2298.01 31.83 6374.76

Expression (22) is drawn in Fig. 7 with the observed data curve. Each trace has been recorded by exchanging 360 thousands packets between both stations. Overflowing buffers in at the AP is the main contribution to lost packets. Hence, burst distribution is dependent on the total network traffic.

0.8

0.6

0.4

0.2 200 150 100 50 RTP packet rate (pps) 0 0 200 bit rate (kbps) 400 600

We have assumed I25 (see Fig. 3 and 4) since longer lengths rarely occur. Besides, in chi-square test, each interval must have more than 5 observations, Nn>5. In (20), the lower Z is, the better fitting is achieved. Table III shows Z for both pmfs, geometric (GE) and logarithmic (Proposed) series. For instance, in Fig. 3 and 4, our model provided Z=1074.71 and Z=335.24 whereas Z values for GE were 2074.45 and 10231.49, respectively. Although our hypothesis was as rejected in 86.1% of all cases, the proposed model was 63.9% superior to GE. This percentage value is determined by dividing the number of cases that the proposed model fits better than GE (equal to 23) by the total number of cases (equal to 36, see Table III).

Fig. 6.

curve in Fig. 7 correspond to half Equation (22) and the of the total source traffic. For instance, if the total network traffic is only produced by one station delivering 200 value is obtained by substituting PR in (22) packets/s, the

by 100. Expression (22) is then valid for the total source traffic between 50 and 400 packets/s. We have considered a total interfering network traffic of 12.2 Mbps exchanged via the same AP.

1

Despite data rates of 4 kbps and 13 kbps are typical rates of compressed digital voice, 256 kbps is only one of several possible data rates for digital video transmission. Therefore, we extended our model, more precisely scenario S21, to cope other video bit rates. Truly speaking, it would be interesting to in function of the bit rate exchanged between both find out stations Sta1 and Sta2. In addition to 256 kbps, we setup four more S2x scenarios at 32, 64, 128 and 512 kbps. From all collected data and concerning the error-free gap length is almost distribution defined by (6) with Y replacing X,

0.9

0.8

0.7

0.6

0.5

values of the constant and equal to 0.9983. Figure 6 shows burst length distribution, P[X]. is Observing Fig. 6, we concluded that the value of

independent of the bit rate. It actually dependents only on the , averaging packet rate (PR). Thus, the best fitting curve for over the five bit rates on both stations, gives,

0.4 40 60 80 100 120 140 RTP packet rate (pps) 160 180 200

VII. CONCLUSIONS

This research work focuses on modeling of multimedia packet loss and packet arrival in 802.11g environment. We have compared our model to the first order Markov GE model and proved that for IEEE 802.11g network, the proposed model is

(22)

814

IEEE Transactions on Consumer Electronics, Vol. 51, No. 3, AUGUST 2005 [19] L. Wilhelmsson and L. Milstein, On the effect of imperfect interleaving for the Gilbert-Elliott channel, IEEE Trans. Com., vol. 47, N. 5, pp. 681-688, May 1999. [20] H. Schulzrinne, S. Casner, R. Frederick, V. Jacobson, RTP: A transport protocol for real time applications, IETF, RFC 1889, Jan 1996. [21] J. Neter, W. Wasserman, and G. Whitmore, Applied Statistics, Allyn and Bacon, 1992, pp. 482-492. Luis Carvalho graduated in Systems and Computer Engineering at Minho University, Portugal, in 2002. Currently, he is a MSc student at Aveiro University. Since 2002 he has been a Research Assistant at the Telecommunications Institute, Portugal. His major interests are on Interactive Television and MPEG-4. Luis has actively been involved in the European Union project FP5-GMF4ITV where he was in charge of the MPEG-2 MUX/DEMUX, rate/frame/picture size transcoding, synchronization and encapsulation. Joao Angeja graduated in Electronics and Telecommunications Engineering at Aveiro University, Portugal, in 2002. He is currently working towards his MSc degree at Aveiro University. Since 2002 he has been research Assistant at the Telecommunications Institute, Portugal. His major interests are on networking with particular emphasis on real time packet transmission and multimedia session control protocols. Joao has been involved in the design of a wireless home platform where he was in charge of researching a new wireless indoor network model. Recently, in a project supported by the Portuguese Navy, he implemented the NATO STANAG 5066 and all network protocols (RTP/UDP/IP). Antonio Navarro (S89-M97) was born in Mozambique in 1966. He graduated (five years first degree) in electrical engineering from Coimbra University, Portugal in 1989 and received the MSc and PhD degrees from the University of Coimbra, Portugal and the University of Newcastle, UK in 1993 and 1996, respectively. He is currently Professor at the Electronics and Telecommunications Engineering Department at Aveiro University, Portugal, where he lectures courses on coding and transmission of video/TV signals. In the 2nd semester of 2004, he was on sabbatical leave at University of Southern California-USA. His research interests are on information theory, optimization, rate-distortion, digital television, video coding, video scalability and transcoding, multiple description coding and reliable wireless transmission of video based multimedia services. Besides theoretical work, Prof. Navarro has supported national and international companies through industrial cooperation projects. He has been the leader of several prototype developments, as for instance, a multimode digital TV Set-Top-Box in 2002, an audio visual coding and HF transmission system in 2004, and a distributed wireless home DTV platform in 2004. In 1990, in the early years of video coding, he co-designed and implemented an ITU-T H.261 modular solution with fourteen TMS320C30 DSPs and one year later with a LSI Logic chip set. He is the Head of the Digital Television and Mobile Video (DTMV) research group at the Institute of Telecommunications (IT), a group with a long experience, about 20 years, in researching and developing in the area of video coding and transmission. He has been involved in some DVB TMs and in some MPEG AdGs, served as a reviewer of several IEEE journals and conferences as well as acted as a consultant to the Portuguese Frequency Regulatory Body in activities of digital terrestrial TV. He is also a project evaluator of proposals submitted to the Portuguese national foundation (AdI) which is in charge of cooperation projects between the industry and the academia including the European Eureka initiative on telecommunications, Celtic. Prof. Navarro has participated in more than 20 national and European projects and co-authored over 60 papers and one patent. He has a solid experience on visual coding, networking and wireless transmission. Recently, in 2005, he received the IT Scientific Award.

1.77 times more accurate than GE. Besides, we showed that the communication performance is independent of the bit rate, in a range of total source traffic between 64 kbps and 1 Mbps. Instead, the packet rate is the only factor that influences the proposed model parameter, in a logarithmic manner. We should mention that this conclusion was achieved with an interference of 12.2 Mbps in the network. Despite our model has only been tested on 802.11g, it could also model other communication networks. Our model can be used to test the robustness of compressed image and video encoded using any recently issued algorithm or standard, such as JPEG 2000 Part 11, ITU-T H.264 or scalable video codecs (SVC). Our main contribution is indeed a concise, efficient and adequate representation of run and burst processes of multimedia packets over wideband/broadband wireless channels.

REFERENCES

[1] [2] [3] [4] [5] [6] [7] [8] [9] [10] [11] [12] [13] [14] [15] [16] [17] [18] E. N. Gilbert, Capacity of a burst-noise channel, Bell Syst. Tech. J., vol. 39, pp. 1253-1265, Sept. 1960. E. O. Elliott, Estimates of error rates for codes on burst-noise channels, Bell Syst. Tech. J., vol. 42, pp. 1977-1997, Sept. 1963. IEEE 802.11g Part 11, Wireless LAN Medium Access Control (MAC) and Physical Layer (PHY) specifications: Further higher data rate extension in the 2.4 GHz band, IEEE, 2003. J. E. Yee and E. J. Weldon, Evaluation of the performance of errorcorrecting codes on a Gilbert channel, IEEE Trans. Com., vol. 43, N. 8, pp. 2316-2323, Nov. 1989. M. Mushkin and I. Bar-David, Capacity and coding for the GilbertElliott channels, IEEE Trans. on Inform. Theory, vol. 35, pp. 12771290, Nov. 1989. J. Garcia-Frias and J. Villasenor, Turbo decoding of Gilbert-Elliott channels, IEEE Trans. Com., vol. 50, N. 3, pp. 357-363, Mar. 2002. P. Frossard and O. Verscheure, Joint Source/FEC rate selection for quality-Optimal MPEG-2 video delivery, IEEE Trans. on Image Processing, vol. 10, N. 12, pp. 1815-1825, Dec. 2001. K. Stuhlmuller, N. Farber, M. Link, and B. Girod, Analysis of video transmission over lossy channels, IEEE JSAC, vol. 18, N. 6, pp. 10121032, Jun. 2000. ISO/IEC 13818-2, ITU-T Rec. H.262, Generic coding of moving pictures and associated audio information: Video, ISO/IEC, 1995. ITU-T Rec. H.263, Video coding for low bitrate communication, 1998. S. Gnavi, A. Grangetto, E. Magli, and G. Olmo, Rate allocation for video transmission over lossy correlated networks, Electronics Letters, vol. 38, N. 20, pp. 1171-1172, Sept. 2002. ISO/IEC 14496-10, ITU-T Rec. H.264, Advanced video coding for generic audiovisual services, May 2004. M. Grangetto, E. Magli, and G. Olmo, Ensuring quality of service for image transmission: Hybrid loss protection, IEEE Trans. on Image Processing, vol. 13, N. 6, pp. 751-757, Jun. 2004. A. Said and W. Pearlman, A new, fast, and efficient image codec based on set partitioning in hierarchical trees, IEEE Trans. on Circuits Syst. Video Technol., vol. 6, pp. 243-250, Jun. 1996. A. Skodas, C. Christopoulos and T. Ebrahimi, The JPEG 2000 still image compression standard, IEEE Signal Processing Mag., vol 18, pp. 36-58, Sept. 2001. H. Wang and N. Moayeri, Finite-state Markov channel-A useful model for radio communications channels, IEEE Trans. on Veh. Technol., vol. 44, N. 1, pp. 163-171, Feb. 1995. A. Kopke, A. Willig, and H. Karl, Chaotic maps as parsimonious bit error models of wireless channels, in Proc. IEEE INFOCOM, San Francisco, CA, Mar. 2003, pp. 513-523. IEEE 802.11b Part 11 Wireless LAN Medium Access Control (MAC) and Physical Layer (PHY) specifications: Higher-speed physical layer extension in the 2.4 GHz band, IEEE, 1999.

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