A Multiplex-Multicast Scheme that Improves System Capacity of Voice-over-IP on Wireless LAN by 100% *

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1 A Multiplex-Multicast Scheme that Improves System Capacity of Voice-over-IP on Wireless LAN by 100% * Wei Wang, Soung C. Liew, Qixiang Pang Department of Information Engineering The Chinese University of Hong Kong Shatin, N.T., Hong Kong {wwang2, soung, Abstract Voice-over-IP (VoIP) is an important application on the Internet. With the emergence of WLAN technology and its various advantages compared with the traditional wired LAN, it is fast becoming the last-mile of choice for the overall Internet infrastructure. This paper considers the support of VoIP over b WLAN. We show that although the raw WLAN capacity can potentially support more than 500 VoIP sessions, various overheads bring this down to only 12 VoIP sessions when using GSM 6.10 codec. We propose a novel multiplexing scheme for VoIP which exploits multicasting over WLAN for the downlink VoIP traffic. This scheme can achieve a 100% improvement in system capacity for b, a, and g. In addition, we present results showing that the delay and delay jitter introduced by the proposed scheme are small. We believe that the scheme can reduce the blocking probability of VoIP sessions in an enterprise WLAN significantly. Keywords VoIP; Wireless LAN; ; Multicasting; Multiplexing; Internet Telephone; Wireless Communiations I. INTRODUCTION Internet telephony is one of the fastest growing applications for the Internet today [1]. The tremendous growth of Internet telephony is driven by a number of its fundamental benefits over traditional public switched telephone network (PSTN). One benefit is the substantial cost reduction since Internet telephony can bypass international toll charges imposed by telecommunication companies. Another benefit is that it enables the creation of a new class of services that combine voice communication with other media and data applications like video, white boarding and file sharing. Additionally, from the network operator s point of view, IP telephony can dramatically improve bandwidth efficiency by exploiting advanced voice compression techniques and bandwidth sharing of packet switching [5]. At the same time, driven by huge demands for mobile access, the wireless LAN (WLAN) market is taking off * This work was sponsored by the Area of Excellence in Information Technology of Hong Kong. quickly. Because of its convenience, mobility, and highspeed access, WLAN represents the future trend for last-mile Internet access. The current most popular WLAN standard, IEEE b, can support data rates up to 11Mbps. Typical bandwidth requirements for a Voice over IP (VoIP) stream are less than 10Kbps. Based on a simplistic computation, one may derive that the number of simultaneous VoIP streams a WLAN can support is around 11M/10K = 1100, which corresponds to about 550 VoIP sessions, each with two VoIP streams. Table 3 shows that the maximum throughput of a WLAN is around 5 to 6 Mbps after various overheads are taken into account. One may be tempted to conclude that more than 100 VoIP sessions can therefore be supported while leaving much bandwidth for other types of traffic. It turns out that the current WLAN can only support about 12 VoIP sessions, a far cry from the above estimate. This surprising result is mainly due to the added packet-header overheads as the short VoIP packets traverse the various layers of the standard protocol stack, as well as inefficiency inherent in the WLAN MAC protocol. For large IP packets of 1000 bytes, the transmission time for the payload at 11Mbps is 1000 * 8 / 11 = 727 µsec. But the total transmission time for the whole MAC frame is 1562 µsec. The additional overhead of more than 800 µsec causes the efficiency at the MAC layer to drop to 46.5%. For VoIP, the drop of efficiency is much worse. A typical VoIP packet at the IP layer consists of a header of 40 bytes and a payload of bytes, for a total of bytes. The transmission time for this VoIP packet in the WLAN is only µsec. The additional overhead of more than 800 µsec overwhelms the total transmission time, nullifying the benefit of voicecompression techniques that enables small VoIP payload in the first place. Note that gathering more data bytes of a voice stream into a VoIP packet before transmission does not work because this will increase the overall endto-end delay of the voice conversation.

2 This paper proposes a voice multiplexing scheme to overcome the large overhead effect of VoIP in WLAN. Our scheme makes use of the features of the multicast mode of WLAN. We will show that the number of VoIP sessions that can be supported can be doubled with this simple technique, while maintaining small delay. II. BACKGROUND A. VoIP For VoIP, the analog or PCM voice signals are encoded and compressed into a low-rate packet stream by codecs. Table 1 lists the attributes of several commonly used codecs in Internet telephony. Generally, the codecs generate constant bit-rate audio frames that have a 40-byte IP/UDP/RTP header followed by a relatively small payload. In this paper, we focus on the GSM 6.10 codec in our discussion, although the general principle we propose is applicable to other codecs as well. For GSM 6.10, the payload is 33 bytes. The time between two adjacent frames is 20 ms, corresponding to 50 packets per second. B. IEEE There are two access mechanisms specified in the IEEE standard: Distributed Coordination Function (DCF) and Point Coordination Function (PCF). PCF is a centralized mechanism, where one central coordinator polls stations and allows them undisturbed, contention free access to the channel. However, PCF is an option not supported in most commercial products. DCF is based on the Carrier Sense Multiple Access with Collision Avoidance (CSMA/CA) protocol. The basic operation of DCF is described in Fig. 1. Before transmission, a station will randomly choose a backoff time with number of time slots ranging from 0 to Contention Window (CW) -1. The station will decrease the backoff-timer counter progressively while the channel is idle for a DCF Inter Frame Space (DIFS) and pause the timer if it senses the channel busy (i.e., if another station is transmitting). When the backoff value reaches zero, the station will transmit its packet. Figure 1. The basic operation of DCF TABLE I.ATTRIBUTES OF COMMONLY USED CODECS Codec Bit rate (Kbps) Framing interval (ms) Payload (Bytes) GSM 6.10 G.711 G G G / / Packets /sec * * For all codecs except G.729, Packets/sec = 1 / (Framing interval). For G.729, two frames are combined into one packet so that Packets/sec = 1/(2* Framing interval) TABLE II.THE PARAMETERS OF B DCF DIFS 50 µsec SIFS 10 µsec Slot Time 20 µsec CWmin 31 CWmax 1023 Data Rate 1, 2, 5.5, 11 Mbps Basic Rate 2 Mbps PHY header* 192 µsec MAC header 34 bytes ACK* 248 µsec * PHY header is transmitted at 1 Mbps, ACK is transmitted at basic rate, 2 Mbps. So they have the fixed transmission time regardless of the data rate. If this is a unicast packet, the station will wait for the receiver to send back an ACK frame after a Short Inter Frame Space (SIFS) interval. If it does not receive the ACK, the station assumes the packet was lost due to transmission error or collision. Then it doubles the CW value, generates a backoff time chosen randomly from the interval [0, CW-1], and retransmits this packet following the same procedure as above. If this is a multicast or broadcast packet, the transmitting station will not wait for the ACK, as multicast receivers do not send back ACKs in general. There are no retransmissions for multicast and broadcast packets in DCF. The station will proceed to multicast/broadcast the next packet regardless of whether the earlier packets have been received successfully or not. In addition, although the maximum radio rate for b is 11Mbps, we found that some commercial products (e.g., Lucent Orinoco, Cisco) transmit multicast packet at 2Mbps bit-rate by default. This is due to the nature that in multicasting, the transmitter does not know who the receivers are (in the common multicasting

3 TABLE III.THEORETICAL AND EXPERIMENTAL THROUGHPUTS UNDER SATURATION CONDITION UDP Payload Throughput under 11 Mbps data rate (Mbps) Throughput under 5.5 Mbps data rate (Mbps) Throughput under 2 Mbps data rate (Mbps) (Bytes) Analysis Experiment Analysis Experiment Analysis Experiment paradigm, the receivers are free to join or leave a multicast session without informing the sender). So, for backward compatibility, the sender uses 2 Mbps to transmit multicast packets so that the earlier versions of products whose maximum data rate is 2 Mbps can receive them. There is usually a flag in the products to control this backward compatibility. We can simply disable this flag to use 11 Mbps multicast. The values of the parameters of b DCF are listed in Table 2. From these parameters, we can get the theoretical maximum throughputs from the relationship: Throughput = Data rate * (Payload / (Payload + Overhead)). The theoretical unicast throughputs, assuming payload size of 500, 1000 and 1400 bytes, (typical IP packet size) are listed along with our experimental throughputs in Table 3. We can see that when the data rate (raw radio transmission rate) increases, the efficiency of decreases. This is because larger data rates will shorten the transmission time for payload, but the transmission time for the overhead changes little, so the effect of the overhead becomes larger. C. Related Work Our idea of multiplexing multiple voice streams in VoIP application is inspired by previous work. Sze et al. [2] proposed a multiplexing system with RTP header compression to reduce VoIP traffic. The focus of the work was on achieving high bandwidth efficiency in core or wide area networks rather than in LAN or WLAN. Our paper shows how to adapt the scheme to make it suitable for use in LAN/WLAN using multicasting. There has been previous work to improve the VoIP capacity on WLAN [3][4]. But their schemes focus on how to modify the standard to work well for VoIP. Our scheme, however, does not attempt to change the current standard and is compatible with today s products. III. VOIP MULTIPLEXING AND MULTICASTING IN WLAN A. System Architecture The basic building block of an network is the basic service set (BSS). There are two types of BSSs: Independent BSS and Infrastructure BSS. Stations in an independent BSS communicate directly with each other and thus must be within direct communication range. Each independent BSS consists of a small number of stations set up for a specific purpose and for a short period of time. For that reason, independent BSSs are often referred to as ad hoc networks. In contrast, stations in an infrastructure BSS communicate with each other via an access point (AP). That is, all traffic to and from a station must flow through the AP, which acts as a base station. APs are in turn connected to each other directly or indirectly (via router, etc.) to allow stations in different BSSs to communicate with each other. This paper focuses on infrastructure BSSs. We assume that all voice streams are between stations in different BSSs, since users seldom call their neighbors in the same BSS. All voice traffic generated within a BSS is delivered to their called parties located at another BSS. For illustration, let us consider the network architecture as shown in Fig. 2a. Each AP has two interfaces, an interface which is used to communicate with wireless stations, and an Ethernet interface which is connected to the voice gateway. Two gateways for different BSSs are connected through the Internet. The voice gateway is required by the H.323 standard and is used for address translation, call routing for signaling and admission control purposes. All voice packets will go through the gateway before entering the WLAN. In the subsequent discussion, we will assume that our proposed voice multiplexer resides in the voice gateway. This is purely for the sake of having a concrete reference design for us to expound on the multiplex-multicast concept. In general, the functionality of the voice multiplexer could reside in the voice gateway, a specially-designed AP, or a server between the voice gateway and a general-purpose AP. In one BSS, there are two streams for each VoIP session. The uplink stream is for voice originating from the station to the AP. The downlink stream is for voice originating from the other side of the VoIP session to the station, which flows from the remote gateway to the local gateway, and then through the AP to the station. B. Packet Multiplexing The main idea of our schemes is to combine the data from several downlink streams into a single packet for

4 transmission on the WLAN so that they can share the overhead. Figure 2a. Network topology and original traffic Figure 2b. Traffic with VoIP multiplex-multicast scheme Furthermore, since the link from the AP to the stations is the last hop, we can simply use multicast to transmit the multiplexed packet to all stations. The MUX and DEMUX procedures are illustrated in Fig. 3. Specifically, the downlink VoIP traffic first goes through a multiplexer (MUX) in the voice gateway. The MUX replaces the RTP, UDP and IP header of each voice packet with a compressed miniheader, combines multiple packets into a single multiplexed packet, then multicasts the multiplexed packet to the WLAN through the AP using a multicast IP address. All VoIP stations are set to be able to receive the packets on this multicast channel. The payload of each VoIP packet is preceded by a mini-header in which there is an ID used to identify the session of the VoIP packet. The receiver to which the VoIP packet is targeted makes use of this ID to extract the VoIP packet out of the multiplexed packet. The extraction is performed by a demultiplexer (DEMUX) at the receiver. After retrieving the VoIP payload, the DEMUX then restores the original RTP header and necessary destination information, and assembles the data into its original form before forwarding it to the VoIP application. Other details of context mapping can be found in [2]. All the stations will use the normal unicasting to transmit uplink streams. The AP delivers the upstream packets it receives to the other BSS, whereupon the voice gateway at the other BSS sends the packets to their destinations using the same multiplexing scheme described above. From Fig. 2b, we see that this scheme can reduce the number of VoIP streams in one BSS from 2 n to n + 1, where n is the number of VoIP sessions. The MUX sends out a multiplexed packet every T ms, which is equal to or shorter than the VoIP inter-packet interval. For GSM 6.10, the inter-packet interval is 20 ms. Larger values of T can improve bandwidth efficiency since more packets can be multiplexed, but the delay incurred will also be larger. For example, if T = 10 ms, every two multiplexed packet contains one voice packet from a VoIP stream. The maximum multiplexing time for one voice packet is 10 ms. If T = 20 ms, every multiplexed packet will contain one voice packet from a VoIP stream, and the maximum multiplexing time is 20 ms. By adjusting T, one can control the tradeoff between bandwidth efficiency and delay. C. Header Compression Besides aggregating VoIP streams, we can also increase the bandwidth efficiency by compressing the packet headers during multiplexing. The idea of RTP/UDP/IP header compression comes from two properties in most types of RTP streams. The first is that most of the fields in the IP, UDP and RTP headers do not change over the lifetime of an RTP session. Second, RTP header fields like sequence number and timestamp are increased by a constant amount for successive packets in a stream. So differential coding can be applied to compress these fields into fewer bits. Our compression is similar to the scheme proposed in [2]. It depends on the use of context-mapping tables in MUX and DEMUX to record necessary information such as RTP header for future reconstruction, source IP address for differentiation between VoIP sessions, synchronization for proper (de)compression and (de)multiplexing. With this scheme, the RTP+UDP+IP header can be replaced with a 2-byte miniheader for most voice packets. We refer the reader to [2] for details. The major reason for the improved efficiency of our system here is the MUX/DEMUX scheme rather than the header compression scheme. IV. PERFORMANCE A. Capacity Analysis Let n be the maximum number of sessions that can be supported. The transmission times for downlink and uplink packets are T down and T up, respectively. Let Tavg be the average time between the transmissions of two consecutive packets in a WLAN. That is, in one second, there are totally 1/ Tavg packets transmitted by the AP and all the stations. So, 1 / T avg = number of streams * number of packes sent by one stream. (1)

5 1)The original VoIP on WLAN We have n downlink unicast streams and n uplink unicast streams. Using GSM 6.10, the payload is 33 bytes. RTP + UDP + IP + MAC = 74 bytes. For both the downlink and uplink traffic packet, the constant overhead is DIFS + average CW + PHY = = 552 usec. In addition, SIFS + ACK = 258 usec. We have ignored the possibility of collisions and the increase of backoff time in subsequent retransmissions after a collision in the analysis here. Our experiments show that the additional overhead caused by collision is small compared with the overheads considered here. Assuming 11Mbps radio rate, T T = ( ) * 8 / = 889 down = up usec On average, for every downlink packet, there is a corresponding uplink packet. So, Tdown + Tup Tavg = = 889 usec 2 Each GSM 6.10 stream has 50 packets per second. From (1), we have 1/ T avg = 2n * 50 Solving this equation, we get n = We see that WLAN can only support around 11 VoIP sessions. 2)VoIP with multiplex-multicast scheme on WLAN The RTP+UDP+IP header of every unmultiplexed packet is compressed to 2 bytes. UDP + IP + MAC = 62 bytes. = ( ) * n + 62 *8 /11+ usec T down [ ] 552 Figure 3. MUX/DEMUX Procedure Tup = 889 usec. Here on average, for one downlink packet, there are totally n corresponding uplink packets. So, Tavg = ( Tdown + n * Tup ) /( n + 1) Solving the above three equations with 1 / T avg = ( n + 1) * 50 we get n = We also derive the capacities when other codecs than GSM 6.10 are used in a similar way, and the results are listed in Table 5. We see that for most of the codecs, our scheme can nearly double the capacity. Note that in the above, we assume the average CW wait time to be 15.5 time slots (i.e., (CWmin-1)/2). When there is more than one station, the average CW wait time is in fact smaller than this. This accounts for the observation in our simulations (see Table 4) that the maximum session is actually a little bit larger. B. Experimental and Simulation Results We have validated our capacity analysis for the original VoIP scheme over WLAN through network experiments and simulations. We have also validated our multiplex-multicast scheme with simulations. The simulator ns-2 [11] is used. The setups for experiments and simulations are the same. We only consider the local part (BSS1 plus voice gateway) of the network shown in Fig. 2a, since our focus is on WLAN, not the Internet. We use CBR as our VoIP traffic model. The packet size and frame generation interval follows the GSM 6.10 codec. In the experiments and simulations, we increase the number of VoIP sessions until the per stream packet loss rate exceeds 1%. We define the system capacity to be

6 the number of VoIP sessions that can be supported while maintaining the packet loss rate for every stream to be below 1%, which corresponds to a good voice quality. For regular VoIP over WLAN, our experiments and simulations yield the same capacity of 12. One observation is that when there are 12 sessions, the packet loss for all the uplink and downlink streams are below 1%; however, the addition of the 13 th session leads to a large increase of packet loss for the downlink streams to 6.7% while the loss rates for the uplinks are still below 1%. This result is due to the symmetric treatment of all stations in : the AP is no different from other stations as far as the MAC layer is concerned. For VoIP, the AP needs to transmit n times more traffic than each of the other stations. When n is smaller than the system capacity, there is sufficient bandwidth to accommodate all transmissions of the AP. However, when n exceeds the system capacity, there is no extra bandwidth to accommodate the AP. Since all stations are treated the same, and the extra traffic from the AP will be curtailed, leading to a large packet loss rate. This observation provides an alternative explanation as to why our scheme can improve the VoIP capacity. Assuming n VoIP packets are multiplexed into one packet, the AP traffic is reduced to be the same as the traffic of each of the other stations in terms of number of packets per second. The results of our scheme are listed in Table 4. The simulation shows that the capacity can be improved to 24, a 100% improvement over the regular scheme. When the number of sessions is 24, our results show that the downlink loss rate is 0.3% while the uplink loss rate is close to 0%. C. Transmission Errors In the above analysis and simulation, we assume the bit-error-rate is low, few packets are lost due to transmission errors. From our experiment and other paper s observation [10], the channel condition is usually quite good when the wireless stations are in proximity of the AP, and the transmission errors can be ignored. However, if the bit-error rate is high, such as for boundary cases where the wireless stations are far away from the AP, we may need to add error controls in our scheme. In this case, since ARQ is not used for multicasting in WLAN, the downlink may be more susceptible to transmission errors than the uplink. We will leave this as a possible direction for future work. D. Local Delay In the following, we present results of our investigation on the local delay suffered within the WLAN and at the multiplexer. A VoIP stream may stream may traverse a long-distance wide-area network in addition to the local area network. From an end-toend viewpoint, it is essential for the local delay to be small so that the overall end-to-end delay of the VoIP stream can be bounded tightly to achieve good quality of service. TABLE IV.ANALYSIS VS. SIMULATION: CAPACITY OF ORIGINAL VOIP AND MULTIPLEX-MULTICAST SCHEME ASSUMING GSM 6.10 CODEC Different Analysis Simulation Schemes Original VoIP Multiplex- Multicast Scheme TABLE V.CAPACITY ANALYSIS FOR DIFFERENT CODECS Codecs Original VoIP Multiplex- Multicast Scheme GSM G G G G )Access delay Access delay is the delay incurred in the wireless network interface. We define it to be the time between when a packet is released into the interface queue until it is totally transmitted. Thus, the access delay includes the waiting delay at layer 2, contention delay to acquire the wireless channel, and the transmission time of the packet. Fig. 4a shows the delays of the original VoIP for successive packets of a session, assuming the total number of sessions is 12 (i.e., the system capacity). The graph on the left is the access delay incurred within the AP for the downlink traffic, while the graph on the right is the access delay within a wireless station for the uplink traffic. As can be seen, the access delay of the AP is larger than that of a station on average. That is reasonable since the AP has much more traffic than other stations, so its interface queue is more likely to build up when the system capacity is reached, leading to a larger delay. The average delay and jitter for the AP (about 15 ms and 4 ms) and wireless stations (about 7 ms and 5 ms) are acceptable, where we define the delay jitter to be the standard deviation. Instead of delay jitter, we can also

7 look at the probability of access delay being smaller than a small value. For example, assuming an end-to-end delay budget of, say, second for a voice stream, it is reasonable to require the access delay incurred in the Figure 4a Access Delay of AP and Station of Original VoIP on WLAN when capacity of 12 is reached Figure 4b Access Delay of AP and Station of Multiplex-Multicast Scheme when number of VoIP session is also 12 Figure 4c Access Delay of AP and Station of Multiplex-Multicast Scheme when capacity of 24 is reached WLAN to be no more than, say, 0.05 second. Table 6 tabulates this delay distributions, where A stands for the random variable representing the access delay. It shows that the access delay can be bounded quite tightly so that acceptable end-to-end delay can be achieved. Fig. 4b shows the delay after applying our multiplexmulticast scheme when the number of VoIP sessions is also 12. It is obvious that the delay is now drastically decreased compared to that of Fig. 4a. Fig. 4c shows the delay when the capacity of 24 is reached. The average delay and jitter for the AP (about 2 ms and 1 ms) and the wireless stations (about 3 ms and 3 ms) are much less than the original VoIP (see previous paragraph). Note from Fig. 4c that the access delay of the AP is smaller than the access delay of the wireless station. This may be because packet transmission at the AP makes use of multicasting and lost packets are not retransmitted, while the transmission at a wireless station makes use of the regular b protocol in which lost packets are retransmitted a number of times before being discarded. The retransmissions tend to increase the overall average-delay statistics. Just in case the reader thinks that the lack of retransmission may make the loss rate at the AP excessive large for the multicast scheme, we would like to refer back to the result stated in the previous section that the downlink loss rate is only 0.3%. 2)Extra delay incurred by our scheme A voice packet will encounter an extra delay at the MUX when it needs to wait a certain amount of time before the MUX generates a multiplexed packet. Recall that the MUX will send off one multiplexed packet to the AP once every T seconds. Since we set the multiplexing period can be at most one audio-frame period in our study, our scheme ensures that the extra delay incurred by the MUX is bounded by one frame period. Note that only downlink packet goes through the MUX. So the extra end-to-end delay caused by our scheme is also bounded by one stream period (20 ms if using GSM 6.10 codec). To account for the extra delay, we define M to be the random variable representing the extra multiplexing delay for a VoIP packet. We assume M to be uniformly distributed between 0 and 20 ms. Table 7 tabulates the distribution of multiplexing delay incurred in the MUX plus access delay incurred in the AP, and the distribution of access delay incurred at the wireless stations, when our multiplex-multicast scheme is adopted. Compared with Table 6, we see that the delay in our scheme is actually tighter than the delay with the original VoIP scheme, and that the delay budget can be met comfortably. V. CONCLUSIONS This paper has proposed a multiplexing scheme for VoIP over WLAN. Our scheme aggregates downlink voice packets with header compression in the voice gateway, then multicasts the multiplexed packet to all the stations. The scheme can reduce the large overhead when VoIP traffic is delivered over WLAN. By means of analysis and simulation, we have shown that our scheme can increase the system capacity by 100%. Specifically, the numbers of simultaneous twoway VoIP sessions that can be supported with and without our scheme are 24 and 12, respectively, in b. Although not shown in the main body of this paper, we have also performed a back-of-the-envelop calculation for a and g. Again the increase in capacity is about 100%. For a, the capacities with and without our schemes are 100 and 52,

8 respectively. For g, assuming compatibility with b except the higher data rate of 54 Mbps, the capacities are 29 and 15, respectively. TABLE VI.ACCESS DELAY DISTRIBUTION FOR ORIGINAL VOIP WHEN THE SYSTEM CAPACITY OF 12 IS REACHED Access delay for AP Access delay for station Pr[ A 0.01s] Pr[ A 0.03s] Pr[ A 0.05s] simultaneous VoIP sessions with the regular scheme will probably have to be reduced to around 5 for b and g to make room for traffic of other applications. This may result in an unacceptably high blocking probability for VoIP in many situations. Our Multiplex- Multicast scheme is one way to increase the VoIP capacity and decrease the blocking probability. TABLE VII.ACCESS DELAY PLUS EXTRA DELAY DISTRIBUTION FOR THE MULTIPLEX-MULTICAST SCHEME WHEN THE SYSTEM CAPACITY OF 24 IS REACHED Access delay for AP plus Access delay for station Extra delay in the MUX Pr[ M + A 0.01s] Pr[ A 0.01s] Pr[ M + A 0.02s] Pr[ A 0.03s] Pr[ M + A 0.03s] 1 Pr[ A 0.05s] 1 We have also investigated the local delay at the WLAN and the multiplexer used in our proposed scheme. Our results indicate the delays and delay jitters for our multiplex-multicast scheme and the regular VoIP scheme are comparable when the number of VoIP session in the former is 24 and that in the latter is 12. Furthermore, in both cases, the local delay can typically be bounded to below 0.05 second, and is small compared to an end-to-end, one-way delay budget of second. Based on our delay results, we believe that DCF can support VoIP and that PCF may not be necessary, so long as the capacity of the WLAN is not exceeded. In an enterprise environment, where other types of traffic (e.g., TCP) may co-exist with the VoIP traffic, the actual number of VoIP sessions that can be supported will be reduced. It remains to be verified whether the delay results hold up when the VoIP traffic is subjected to interference from other traffic. In a future paper we will address this issue and show how to guarantee the quality of service of different traffic types. In addition, we have assumed good channel conditions in this paper. When the channel is noisy, we envision the need for adding FEC on the multicast downlink for our proposed scheme, and the result will also be presented in a separate paper. What we can conclude from the results of this paper is this. In an enterprise environment, the number of

9 REFERENCES [1] G. Thomsen and Y. Jani, Internet telephony: Going like crazy, IEEE Spectrum, pp , May 2000 [2] Sze, H.P., Liew, S.C., Lee, J.Y.B. and Yip, D.C.S, A multiplexing scheme for H.323 voice-over-ip applications, IEEE J. Select. Areas Commun, Vol. 20, pp , Sept [3] Hung-Huang Liu and Jean-Lien C. Wu, Packet telephony support for the IEEE wireless LAN, IEEE Communications Letters, Vol. 4, No. 9, pp , Sept [4] Rusty O. Bladwin, Nathaniel J. Davis IV, Scott F. Midkiff and Richard A. Raines, Packetized Voice transmission using RT-MAC, a wireless real-time medium access control protocol, Mobile Computing and Communications Review, Vol 5, No.3, pp , July, 2001 [5] Bill Douskalis, IP Telephony, the integration of robust VoIP services, Prentice Hall PTR, [6] Matthew S. Gast, Wireless Networks, the definitive guide, O REILLY, [7] Neeli Prasad and Anand Prasad, WLAN Systems and Wireless IP for Next Generation Communications, Artech House, [8] S. Casner and V. Jacobson, Compressing IP/UDP/RTP headers for low-speed serial links, IETF RFC 2508, [9] T. J. Kostas, M. S. Borella, I. Sidhu, g. M. Schuster, J. Grabiec and J. Mahler, Real-time voice over packet-switched networks,, IEEE Network, vol. 12, No. 1, pp , January/February [10] Philip K. McKinley and Suraj Gaurav, Experimental evaluation of forward error correction on multicast audio streams in wireless LANs, Proceedings of the eighth ACM international conference on Multimedia, pp , [11] The Network Simulator ns2,

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