Addressing the Link Adaptation Problem for VoWLAN using Codec ...

Addressing the Link Adaptation Problem for VoWLAN using Codec Adaptation Philip McGovern, Seán Murphy, and Liam Murphy School of Computer Science and Informatics University College Dublin, Ireland E-mail: [email protected], [email protected], [email protected] Abstract—In this paper a scheme is proposed to deal with congestion problems that can arise due to Link Adaptation (LA) in Wireless Local Area Networks (WLANs) in the presence of VoIP traffic. The proposed scheme operates by first determining if LA has resulted in the system becoming congested. If not, no further action is taken. However, if, LA has resulted in an overloaded system, the voice codec of the handset which has undergone LA is adapted so as to restore the system to its earlier state, thereby alleviating the congestion. This is achieved by specifically adapting the codec so as to maintain approximately the same level of medium usage as was used prior to the LA. The proposed scheme was evaluated on an experimental test-bed and results show that the codec adaptation was very effective at overcoming the problem of LA. Index Terms— Voice over IP (VoIP), IEEE 802.11, Admission Control (AC), Link Adaptation (LA)

I. INTRODUCTION

I

T is now well known that there are significant performance issues with respect to the transport of Voice over IP (VoIP) traffic over IEEE 802.11 Wireless Local Area Networks (WLANs). The small frame sizes encountered in VoIP, coupled with the large overheads associated with the CSMA/CA MAC protocol of 802.11 limits the number of simultaneous VoIP connections to a very small number. Garg and Kappes show that an 802.11b network can only support 6 calls using the G.711 codec with 10ms of voice data per packet, when all terminals use the maximum data rate of 11 Mb/s [1]; with the users transmitting at 1Mb/s, only 2 such calls can be supported. Furthermore, it is known that the onset of congestion in 802.11 is not gradual: the system has a tendency to transition from an uncongested state delivering good performance to a congested state delivering very poor performance with the addition of little extra traffic. In the case of VoIP, the addition of one more call to the system can result in very unacceptable call quality for all users of the system. For these reasons, Call Admission Control (CAC) functionality is a critical requirement of these systems. In a wired environment, once a CAC scheme determines that a call can be admitted, the call can be assumed to receive acceptable quality for its duration. However, in a WLAN environment, even with CAC in place, this is not necessarily the case due to the possibility of Link Adaptation (LA). The support of the Informatics Research initiative of Enterprise Ireland is gratefully acknowledged.

LA is a fundamental component of most wireless systems which can accommodate the variation in channel conditions that can exist on wireless links. The essence of LA is that a wireless node may adapt its modulation and coding scheme (MCS) to the prevailing channel conditions, resulting in more efficient use of the radio resources. This has implications for the rate at which data is transmitted to/from the wireless node. In general, LA can result in transition to a less robust MCS or a more robust MCS, but the focus here is on transitions to more robust MCS as these are more problematic. In the case of WLAN, LA may occur if a mobile user roams away from an Access Point (AP) such that the received signal strength fades and the wireless interface switches to a more robust modulation scheme to maintain an acceptable Bit Error Rate (BER). The more robust modulation scheme and resulting reduction in transmission rate means an increased channel occupancy time to transmit the same amount of data. This increased channel occupancy time may result in the WLAN capacity being exceeded. Thus, LA – like the addition of new calls – may result in congestion on the WLAN and hence unacceptable quality for all existing connections. This paper details a terminal-oriented scheme to deal with the LA problem in a Voice over WLAN (VoWLAN) environment and which can be integrated with a VoWLAN CAC scheme. The scheme operates by adapting the voice codec of a handset which undergoes LA if the drop in transmission rate has caused the WLAN to become congested. The codec is adapted so as to revert back to approximately the same level of medium usage as was used prior to the LA. This measure restores the system to its earlier state, alleviating the congestion and thus enabling all calls to proceed with acceptable call quality. Codec adaptation is based on the choice of two parameters: codec type and packetization interval. An analysis of 802.11 VoIP frame transmissions as detailed in this paper is used in the determination of the parameter choices that will give approximately the same level of medium usage as was used prior to the handset performing LA. Importantly, the scheme does not require modifications to network components such as APs; all that is required is wireless VoIP handsets that implement the proposed scheme. In this work, it is assumed that certain WLAN channels are dedicated to voice only traffic. This is to avoid the coexistence problem of voice and data on the same WLAN cell, which is to be addressed by the recently ratified 802.11e standard. The rest of this paper is organized as follows. In Section II

the background and related work are discussed. The related work includes a summary of a CAC scheme for VoWLAN developed by the authors [2]. In Section III the effects of LA on call quality are discussed. Section IV details how to estimate channel occupancy time for a specified codec and packetization interval. Section V presents the Link Adaptation Codec Adaptation Mechanism (LACAM) and includes some results pertaining to the scheme. Finally, Section VI concludes the paper. II. BACKGROUND A. 802.11 Link Adaptation Wireless channels suffer from various impairments such as path loss, interference, multipath fading and shadowing. A LA algorithm is used by each wireless station to adapt the modulation scheme to the current link conditions to maintain an acceptable BER. 802.11b, for example, supports transmission rates (modulation schemes) of 11Mb/s (CCK), 5.5Mb/s (CCK), 2Mb/s (DQPSK), and 1Mb/s (DBPSK) [3]. LA implementation details were not specified in the standard but most use a variety of link quality measurements such as Packet Error Rate (PER) and received signal strength to determine when to adapt [4]. B. System Architecture and EAC Fig. 1 depicts a system architecture diagram for a VoWLAN phone system based on standard network components. The VoWLAN system can interface with a legacy PBX through a voice gateway as shown. A terminal-oriented CAC scheme for VoWLAN based on the above architecture has been proposed elsewhere by the authors [2] and is summarised here. The CAC scheme is based on the Endpoint Admission Control (EAC) paradigm, with all CAC decisions taken by the endpoint devices. This is a probing based scheme with the probes (ICMP echo packets) resembling the traffic characteristics of the flow to be admitted. The QoS of the probe flow is measured and the flow is admitted if the QoS is acceptable. The scheme focuses on congestion in the wireless access network by probing the path to the VoIP server, as opposed to probing the entire end-to-end path, as this is where the bottleneck typically arises. While the scheme can solve the CAC problem it cannot guarantee acceptable call quality for the duration of a call due

Fig. 1. System architecture diagram for a WLAN phone system.

to the possibility of LA. Hence, it is necessary to augment the above scheme with extra functionality to deal with LA as discussed here. C. Related Work Most CAC schemes for VoWLAN which also address the LA problem are AP-centric, requiring the deployment of specialized network components such as non-standard APs. The majority of these are proprietary, undisclosed solutions. One published solution, [5], focused at the AP calculates the expected channel utilization of the VoIP flow to be admitted. The new flow is admitted if it can be accommodated by the available channel resources as determined by the AP. The LA problem is dealt with by reserving some network capacity which can absorb a change in data rate for some of the connections. This approach to the LA problem can be very wasteful of resources, especially if the capacity reserve is large enough to allow for a user drop in transmission rate from 11Mb/s to 1Mb/s (remembering that 802.11b can only support 2 1Mb/s calls). Standardized solutions for CAC based on the recently ratified 802.11e standard are being proposed [6]. Both the EDCA and HCCA modes of 802.11e provide admission control through the use of a traffic specification (TSPEC). If a node needs more channel access time/transmissions opportunities, due to LA, for example, then it must renegotiate its TSPEC with the AP. If the new TSPEC cannot be accommodated then the TSPEC is rejected and the flow is curtailed; this would typically result in very poor quality for the call on the node which undergoes LA. While 802.11e provides a standardized solution to CAC, there are problems integrating it with the large legacy base of 802.11 a/b/g equipment and hence it is useful to consider alternative approaches. III. EFFECTS OF LINK ADAPTATION ON CALL QUALITY Before discussing the LA experiments it is useful to provide some information on the voice capacity of 802.11b in different configurations. Previous work by the authors investigated the number of VoIP calls (G.711, 10ms packetization, no silence suppression) that could be supported by an 802.11b wireless cell. 802.11b was chosen due to its large install base and G.711 was chosen as the voice codec due to its universal availability. Results showed that 6 11Mb/s calls, 5 5.5Mb/s calls, 3 2Mb/s calls, and 2 1Mb/s calls could be supported by a single 802.11b WLAN [2]. In all cases studied, when the capacity of the cell was exceeded by the addition of an extra call, all calls went from having acceptable access delays (under 5ms) and loss (0%) to having unacceptably high access delays (as high as 240ms) and loss (as high as 24%) in the downlink direction. The bottleneck arises in the downlink because of the well known fairness issues associated with the 802.11 MAC mechanism [1]. All uplink streams still received acceptable quality. Experiments to determine the effects of LA were performed on an experimental testbed. The test-bed consisted of 7 wireless laptops that were associated with an 802.11b AP

operating in DCF mode. The AP connected the wireless network to a wireline network through an Ethernet switch. Four PCs were connected to the wireline network, and acted as endpoints for VoIP connections. The setup was used to make full-duplex Constant Bit Rate (CBR) VoIP calls between a wireless laptop and a wired PC, i.e. each call consisted of two simplex RTP data streams: one for the incoming call leg and one for the outgoing call leg. Software was used to emulate the G.711 codec with 10ms packetization intervals without the use of silence suppression. As the WLAN NIC drivers were not very flexible, the LA had to be emulated. More specifically, the LA was not performed by actually changing the modulation scheme/transmission rate but was simulated by modifying the size of uplink and downlink packet payloads so as to emulate the new lower transmission rate, while maintaining the same actual data transmission rate throughout each individual experiment. One of the laptops and its corresponding wired node ran specific software which enabled them to perform this emulated LA as necessary during each experiment. Clock synchronization between communication nodes was established in order to allow WLAN access delays to be measured. All experiments were carried out in a clean radio environment with all wireless nodes within radio range of one another, i.e. there were no hidden stations. Also, although the WLAN NIC drivers did not explicitly support LA, they did allow the transmission speed to be set manually to any of the supported rates (11 Mb/s, 5.5 Mb/s, 2 Mb/s, or 1Mb/s), so that calls could be added to the system at any of the supported transmission rates. The results presented here focus on the downlink statistics for two reasons. Firstly, the downlink typically acts as a bottleneck and problems usually manifest themselves in the downlink before the uplink. Secondly, as the focus here is on a terminal-oriented solution to the LA problem, the solution must use information available at the handset such as information pertaining to received traffic: this information is most impacted by the WLAN downlink performance. Two sets of experiments were performed. The first set were ‘control’ experiments where the simulated change in transmission rate did not result in the WLAN capacity being exceeded. The second set of experiments accounted for the situation where the simulated change in transmission rate did result in the capacity being exceeded. Each set consisted of individual experiments which represented a different spatial distribution of the wireless clients. Further, in each set of experiments, there were a fixed number of calls which did not adapt their transmission rate and a single call which experienced LA. A substantial number of control experiments were performed to determine the impact of LA which did not cause the WLAN to become congested. In all cases, each call continued to receive acceptable quality after LA. Moreover, the call that underwent LA showed a slight increase in downlink access delay (typically 1-2ms) while the other ongoing calls suffered slightly increased jitter in some cases. This can be seen, for example, in Fig. 2 where there were

initially 3 11Mb/s calls on the system and one experienced LA, reducing its transmission rate to 2Mb/s. The constant increase in access delay is due to the extra time needed to transmit the same amount of data at the new transmission rate. The second set of experiments performed examined the situation where the WLAN capacity was exceeded due to LA. Recalling the voice capacity limits of the system as discussed above enabled it to be brought close to its throughput limit. In these experiments, the LA resulted in unacceptable quality for all ongoing calls. This is reflected in Fig. 3 where the access delays are shown for the case in which there were 5 5.5Mb/s calls on the system and one reduced its transmission rate to 1Mb/s. It can be seen that the downlink access delays increase rapidly to values inappropriate for VoIP and loss ensues. The downlink access delays experienced by the 4 other ongoing calls resemble those of Fig. 3 closely. In all the other experiments carried out similar trends were seen, i.e. the access delays increase quite linearly and then saturate at an unacceptably high value (as high as 160ms in some instances) with the onset of loss (as high as 15%). The observed trends can be explained as follows. When the capacity of a WLAN is exceeded, the AP is curtailed rendering it a bottleneck. This has the effect of causing the downlink delays to begin increasing linearly as the AP’s buffer is filling. When the buffer fills, the delays saturate at an unacceptably high value with the onset of loss which is reasonably evenly distributed across all downlink streams. It was also noted that these effects were not seen on the uplink streams which still receive acceptable service as expected. The insight gained from these experiments was then used in the development of the LACAM to determine when LA has resulted in the onset of congestion. IV. ESTIMATING CHANNEL OCCUPANCY TIME In this section an analysis of VoIP packet transmission in a WLAN environment is detailed. As the focus in this work is on IEEE 802.11b networks due to their large legacy installed base, this analysis pertains to 802.11b transmissions only. The

Fig. 2. Downlink access delays of successive packets for the node which reduced its transmission rate (from 11Mb/s to 2Mb/s) where there were 2 other 11Mb/s calls on the system.

Fig. 3. Downlink access delays of a node whose transmission rate dropped (from 5.5Mb/s to 1Mb/s) with 4 other 5.5Mb/s calls on the system.

analysis provides results which were then used by the handset’s LACAM when performing codec adaptation so as to maintain approximately the same channel occupancy time, as used prior to LA. This analysis is based on the use of voice only (full-duplex CBR) traffic on the WLAN, where each station contends to transmit the uplink call leg while the AP contends on behalf of all active stations to transmit the downlink call legs. Channel occupancy time, Tco, is defined here as the fraction of channel time per unit time required by a full-duplex VoIP call connection for a given codec setting and transmission rate. This metric will allow different codecs with possibly differing packetization intervals to be compared based on their “per second” channel occupancy requirement. The analysis assumes the system is operating close to its capacity limit and ignores time consumed by collisions. The constituent time components of each VoIP packet transmission are shown in Fig. 4. The back-off window size consists of a number of contention slots with each slot lasting 20µs. Following the analysis given in [1], the average back-off window size is assumed to be 8.5 slots, which results when the AP always contends with a single other station. The fixed headers totaling 72 bytes include the MAC header (24 bytes), LLC header (8 bytes), IP header (20 bytes), UDP header (8 bytes), and RTP header (12 bytes). Thus, the channel occupancy time per unit time, Tco, is given by Tco = Tpkt × Npkt × 2, where Npkt is the number of VoIP packets transmitted per second and the factor of ‘2’ is to account for the uplink and downlink streams. Tpkt is the channel occupancy time required per VoIP packet transmission based on Fig. 4 and is given by Tpkt = TDIFS + TBO + 2TPHY + THDR + TP + TFCS + TSIFS + TACK with the individual components as given in Table I. R is the transmission rate of the call connection under consideration and P is the VoIP packet payload in bytes. Table II outlines the channel occupancy time per unit time for the G.711 (64kb/s) and G.729 (8kb/s) voice codecs for 10, 20, 30, and 40ms frame intervals for all possible 802.11b supported data rates with the packet payload sizes as indicated. These are the only codecs considered in this work due to their almost universal availability (i.e. nearly all voice gateways and VoIP clients support these two codecs) however, the approach can also be applied to other codecs. From Table II, for example, it can be seen that an 11Mbps call using G.711 10ms codec has a channel occupancy time of 0.1567s per unit time. If the transmission rate of this call dropped to 1Mb/s, then switching to the G.729 20ms codec would give approximately the same channel occupancy time (i.e. 0.1494s) per unit time as used earlier. V. LINK ADAPTATION CODEC ADAPTATION MECHANISM The terminal-oriented solution to the LA problem

Fig. 4. Breakdown of channel occupancy time for VoIP packet transmission.

TABLE I TIME COMPONENTS OF A VOIP FRAME TRANSMISSION PART

TIME[µS]

TDIFS TBO TPHY THDR TP TFCS TSIFS TACKa

50µs 8.5 × 20µs 192µs 72 × 8 / R P×8/R 4×8/R 10µs 14 × 8 / Rbasic

a For R ε {11, 5.5, 2}Mb/s the ACK is transmitted at the basic rate, Rbasic = 2Mb/s. For R = 1Mb/s the ACK is transmitted at R basic = 1Mb/s.

TABLE II CHANNEL OCCUPANCY TIMES FOR VARIOUS CODEC SETTINGS TRANSMISSION RATE CODEC 5.5MB/S 2MB/S 1MB/S 11MB/S SETTING G.711 10ms 80 byte G.729 10ms 10 byte G.711 20ms 160 byte G.729 20ms 20 byte G.711 30ms 240 byte G.729 30ms 30 byte G.711 40ms 320 byte G.729 40ms 40 byte

0.1567s 0.1465s 0.0842s 0.0740s 0.0600s 0.0498s 0.0479s 0.0377s

0.1794s 0.1590s 0.1013s 0.0810s 0.0753s 0.0549s 0.0623s 0.0419s

0.2588s 0.2028s 0.1614s 0.1054s 0.1289s 0.0729s 0.1127s 0.0567s

0.3948s 0.2828s 0.2614s 0.1494s 0.2169s 0.1049s 0.1947s 0.0827s

highlighted above involves a handset that has just undergone LA determining if this has resulted in the WLAN becoming congested and, if so, adapting its codec. The codec is adapted so as to revert back to approximately the same channel occupancy time as was used prior to the LA. These two important aspects of the mechanism are discussed in the following subsections, followed by two subsections describing parameter selection and obtained results. A. Determination of WLAN Congestion In principle, the terminal can determine if the WLAN has become congested by comparing downlink delays and losses before and after LA occurred. The substantial increase in downlink delays observed when the system becomes congested – as noted in Section III – can be used as the basis of a mechanism that determines when the WLAN is overloaded. However, determining downlink delays is non-trivial: jitter is much more readily available at the terminal and hence the approach that is used here is one in which changes in jitter are used to infer whether increases in downlink delays have occurred. This, however, is not without its problems, largely due to the lack of clock synchronization between endpoints and the unknown end-to-end delays. For this reason, care must be taken when using jitter information to infer increases in delays and, by implication, congestion. The approach described here, based on the notion of Phase Jitter, addresses these issues. 1) Phase Jitter Phase Jitter, J, as defined here, is the difference between the time a packet arrived, A, and the time it was expected, E.

Thus, if the wireless VoIP handset was able to measure J then it would be able to detect an increase in the one-way delay of arriving packets due to congestion at the AP, as any increase in delay results in the same increase in J. So, for each received packet, i, the Phase Jitter can be calculated as, Ji = Ai – Ei. Ai is simply measured by recording the value of the local wall clock at the receiver when the packet arrives. Ei is calculated through mapping the sender’s timeline (as conveyed by the RTP timestamps) to the local timeline as indicated by the receiver’s wall clock. To map from sender’s timeline (i.e. RTP timestamps) to the receiver’s timeline the “relative offset” (RO) between sender and receiver clocks is added to the RTP timestamps. To calculate RO, the receiver tracks the difference, dn, between the RTP timestamp of the nth packet, Tn, and the arrival time of that packet, An: dn = An – Tn. dn includes a constant factor due to the actual clock offset between sender and receiver, a constant factor due to the minimum network transit time, a variable delay due to network queuing jitter, and a rate difference due to clock skew between sender and receiver clocks. Clock skew can be neglected due to the short observation period over which the LACAM operates (typically 1-2s, depending on the codec frame intervals used). The difference is calculated as each packet arrives, and the receiver tracks its minimum observed value to obtain the “relative offset”: RO = min(dn-l, …, dn), where l is the number of observed packets over which RO is determined, and these are the l packets received just prior to a change in transmission rate. The minimum observed value in the sequence dn-l, …, dn corresponds to the packet which suffered the least network queuing delay and as such represents the best estimate of relative offset. Thus, Ji = Ai – Ei = Ai – (Ti + RO). The correlation between phase jitter and downlink delays can be seen by comparing Figs. 3 and 5: increases in downlink delays result in corresponding increases in Phase Jitter. 2) Using Phase Jitter to Detect Congestion The insight gained from the analysis of the results in Section III motivated the development of a delay threshold based approach to congestion detection due to the difference in delays between a congested and an uncongested WLAN. This procedure which uses a moving window as discussed below is only performed by a handset for a short observation period after it detects a change in its transmission rate. Each handset maintains two circular arrays of length l which contain the timing information (RTP timestamps in one and arrival times in the other) of the l most recently received packets. Upon a change in its transmission rate the handset determines the minimum RO over these l packets and then uses this value to calculate the phase jitter of arriving packets. Addition of timing information to the circular arrays is then suspended until the procedure below completes. A moving window of length w which contains the phase jitter values of newly arriving packets (i.e. packets received

Fig. 5. Phase jitter delays of a node whose transmission rate dropped (from 5.5Mb/s to 1Mb/s) with 4 other 5.5Mb/s calls on the system.

after the drop in transmission rate) is then used to determine if codec adaptation is to be performed. w packets must first be received before the first decision is made in order to allow the window to fill. If all w phase jitter values exceed a predetermined threshold, T, then the codec is adapted. If this is not the case then the window is updated with a new phase jitter value upon the arrival of the next packet, and the criterion is again tested. This step is repeated until either the codec is adapted or until N packets have been received subsequent to the handset performing LA, at which point the procedure completes and the codec is either adapted if all w phase jitter values exceed T, else no action is taken, i.e. congestion has not been detected. At this point the circular arrays continue being updated again with the timing information of newly arriving packets. Note that any lost packets are simply counted as packets whose phase jitter exceeded the threshold T. This assumption simplifies the algorithm allowing it to work with a delay only threshold. In summary, by monitoring the phase jitter of packets received after a change in its transmission rate, a handset should be able to determine if its change in transmission rate has caused the onset of network congestion indicated by increasing phase jitter delays and if so, adapt its codec. B. Codec Adaptation If congestion is detected as described above then codec adaptation is triggered, where the new codec setting is chosen based on table lookup. A list of codecs (with possible configurations for each) that may be used dynamically during the call can be established at call setup time using the signaling protocol (e.g. SIP/SDP INVITE message). The channel occupancy times for these available codec settings at all the supported 802.11b transmission rates are then determined at call setup, based on the analysis provided in section IV. These are then used to form the lookup table, which defines codec settings which will give approximately the same channel occupancy times for transitions between the different possible transmission rates. If, for certain reasons, the codec cannot be adapted sufficiently (e.g. the other party only supports one codec setting), such that the congestion cannot be alleviated, the signaling protocol is also used to teardown the call connection terminating the call. This measure will restore the quality levels of the other ongoing calls. If the LA does not cause the system to enter a congested state, then no action is taken.

C. Parameter Selection The algorithm uses 4 parameters l, w, T, and N, which were chosen as follows: l = 50, w = 10, T = 12ms, and N = 50. The values chosen were based on experimental observations. The value of T was chosen so as to differentiate between a congested and an uncongested WLAN. LA that does not result in congestion will lead to a slight increase in delay of 1-2ms as discussed earlier. Setting T = 12ms will easily allow differentiation between this situation and the situation where LA does result in congestion. There was a tradeoff in the choice of value for w. A small value for w gives quick decisions thus reducing the duration of the temporary degradation to other calls; but too small a value for w would leave the window susceptible to incorrect decisions. Setting w = 10 was found to perform well in a variety of scenarios. There was also a tradeoff in the choice of N. A large value of N is desirable to detect the situation where the capacity is only just exceeded and the buildup in delays is slow. However, N cannot be too large to avoid the situation where a handset may incorrectly adapt its codec due to detected congestion that may be due to some other transient behaviour, e.g. another user probing the network to ascertain if their call may be admitted. With a value of N = 50 the probability of another user probing the network during those 50 packets (i.e. 500ms, for 10ms packetization) is small, while still allowing the detection of congestion where the buildup in delays is reasonably slow. D. Prototype Evaluation and Results A prototype was built and various experiments, similar to those in Section III, were performed (including ‘control’ experiments) to verify correct functionality of the LACAM. In each case the single call which underwent LA initiated its call using the default G.711 10ms codec setting. Codec adaptation was based on choosing from either: G.711 20ms, G.729 10ms, or G.729 20ms. The new setting chosen was based on which one gave an approximately similar channel occupancy time, at the new data rate compared to that used prior to LA, according to Table II. In most experiments the decision to adapt the codec was made reasonably quickly; 15-20 packets (150-200ms, for 10ms packet interval) after the drop in transmission rate. Fig. 6 for example shows the phase jitter values of successive packets for a call which underwent LA resulting in congestion such that the codec was adapted. Note that the phase jitter values displayed for the new codec setting (G.729 20ms) are for illustration purposes only and were calculated based on an updated value being used for the expected arrival time to allow for the packetization delay increase of the new codec setting: in this case the expected arrival time was increased by 10ms in each case. Fig. 7 shows the corresponding downlink access delays of an ongoing call during the same experiment, where the effects of the congestion are visible, as is the recovery of the call. In general the temporary degradation of other ongoing calls lasts between 300 – 600ms where the increase in delays during this period is not too great (typically 40 – 60ms) and could be easily accommodated by an adaptive jitter buffer at the receiver.

VI. CONCLUSION A scheme to deal with LA in a VoWLAN environment was proposed, and which can be used in conjunction with a CAC scheme. The scheme, implemented at each wireless handset, operates by adapting the voice codec of a handset that undergoes LA which causes the WLAN to become congested. This measure deals effectively with the LA problem allowing all ongoing calls to proceed with acceptable call quality, and results show that the period of degradation to the calls is short. While the focus here was on 802.11b, this scheme should

Fig. 6. Phase jitter delays of successive packets for the node which reduced its transmission rate (from 11Mb/s to 1Mb/s) where there were 5 other 11Mb/s calls on the system. The codec was changed from G.711 10ms to G.729 20ms.

Fig. 7. Downlink access delays of an ongoing call showing the temporary degradation due to another call undergoing LA, and the recovery as the other call adapted its codec.

scale naturally to the higher rate 802.11 a/g standards. Also, while the focus was on the use of standard codecs (i.e. G.711, G.729) the scheme can be used with other codecs. REFERENCES [1] [2] [3] [4] [5] [6]

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