Adapting WLAN MAC Parameters to Enhance VoIP Call Capacity

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Adapting WLAN MAC Parameters to Enhance VoIP Call

Capacity

Gr ´ainne Hanley, Se ´an Murphy and Liam Murphy

Dept. of Computer Science, University College Dublin Belfield, Dublin 4, Ireland

hanleyg@gmail.com, sean.murphy@iname.com, liam.murphy@ucd.ie

ABSTRACT

This work describes a detailed simulation-based study of the performance of an IEEE 802.11e Medium Access

Con-trol (MAC) layer over an IEEE 802.11g Physical (PHY)

layer. The study focuses on the number of simultaneous bidirectional G.711 Voice over IP (VoIP) calls that can be supported by a Wireless Local Area Network (WLAN) us-ing the Extended Rate PHY - Orthogonal Frequency Division

Modulation (ERP-OFDM) mode of 802.11g.

A new scheme for adapting the WLAN MAC parame-ters is proposed in this work. The new scheme dynamically adapts the Contention Window (CW) based on the retrans-mission rate of the system. In addition, an adaptive

Trans-mission Opportunity (TXOP) mechanism aids in balancing

the uplink and downlink traffic levels and so provides the equality in uplink and downlink performance that is required for bidirectional VoIP traffic. The proposed scheme can thus maintain acceptable levels of QoS for higher call capacities, increasing the overall VoIP capacity of the system.

Categories and Subject Descriptors: C.2.1 [Network Architecture and Design]: Wireless communication

General Terms: Performance.

Keywords: Wireless LAN, VoIP, Medium Access Control, Quality of Service, Parameter Adaptation.

1.

INTRODUCTION

Quality of Service of VoIPoW systems is currently an area receiving much interest in both the research community and the marketplace. The real-time nature of VoIP means that it imposes strict loss and delay bounds on the network. Since IEEE 802.11-based WLANs have difficulty meeting such strict loss and delay constraints, a key question relates to how best to configure the parameters of the WLAN to deliver the best performance.

In this work, an approach to adapting the WLAN pa-rameters for bidirectional VoIP is proposed. Unlike other schemes that have been proposed, this integrated approach

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°ACM, 2005. This is the author’s version of the work. It is posted here by permission of ACM for your personal use. Not for redistribution. The definitive version will appear in the conference below.

To appear in MSWiM’05,October 10–13, 2005, Montreal, Que-bec, Canada.

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enables two of the key WLAN performance parameters to be adapted simultaneously in order to deliver the best perfor-mance for VoIP traffic. The approach is based on measuring two quantities at the AP - the amount of retransmissions in the system and the uplink and downlink throughput - and adapting the key WLAN parameters appropriately.

The simulations studied here use an IEEE 802.11e MAC layer in association with an IEEE 802.11g PHY layer. The focus here is on the 802.11e distributed mode of operation, called Enhanced Distributed Channel Access (EDCA). Choi et al. described this scheme in detail in [9] with some em-phasis on the packet bursting mechanism – often referred to as Contention Free Bursting (CFB) – which was incorpo-rated into the IEEE 802.11e MAC in order to enhance the system performance.

The focus of this study is a WLAN carrying purely VoIP packets. Such traffic is transmitted using the highest priority

Access Category (AC) in 802.11e.

IEEE 802.11g is an extension of the IEEE 802.11 and 802.11b PHY standards for the 2.4GHz ISM band. The standard outlines 4 transmission schemes but only the ERP-OFDM mode is considered in this work, as it is the only mandatory mode that provides data rates of up to 54Mb/s. This paper is structured as follows. Section 2 outlines some of the published work related to this study. In Section 3, the simulation setup is described in detail. A discussion of the proposed adaptive scheme and the results are presented and discussed in Section 4. Finally, the paper is concluded in Section 5.

2.

RELATED WORK

There has been much activity in the area of WLAN per-formance analysis in the last few years. The most relevant related works are highlighted here.

Bianchi did seminal work in this area by developing the first analytical model [2] to compute the saturation through-put of the Distributed Coordination Function (DCF) scheme. He highlighted the dependence of system performance on the number of nodes in the system and the size of the minimum CW – issues that are very relevant in the scenario studied here.

Other 802.11-related research has focused on approaches to adapting system parameters. For instance, work carried out by Aad et al. [1], showed that the standard CW mecha-nism was quite inefficient. Their results demonstrated that since the number of contending stations often remains rea-sonably stable, resetting the CW to the minimum value after a successful transmission leads to an increase in collisions

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and thus delays. Hence, they proposed a scheme that in-volves a more gradual decrease in the CW value after a suc-cessful transmission. These ideas influenced the approach described here as it consists of a CW which does not change so rapidly.

One other aspect of WLAN behaviour which is impor-tant in this work is that of the disparity between uplink and downlink performance. This was reported by Casetti & Chiasserini in [4], but their work was related to the IEEE 802.11e MAC over the IEEE 802.11b PHY layer. They sug-gested a combined AIFS and CW strategy, rather than the use of TXOP adjustments, to help alleviate the disparity. Pong and Moors [10] demonstrated the advantage of using TXOPs in achieving higher network capacity; use of TXOP is a key aspect of the scheme proposed here.

3.

THE SIMULATION SCENARIO

For this work, a simulation model of IEEE 802.11e using the EDCA scheme and including the CFB mechanism, was used. It was developed by the TKN group in the Technical University of Berlin, as an extension of the Network Simula-tor package, NS-2. Using this model, a series of simulations were performed in order to determine the performance of VoIP in an 802.11e/802.11g system. The standard values for both the 802.11e EDCA scheme and the 802.11g ERP-OFDM PHY parameters were used.

The maximum mandatory data rate of 24Mb/s was used throughout the simulations and – in accordance with the 802.11g standard – the short Physical Layer Convergence

Protocol (PLCP) preamble and short PLCP header were

transmitted at 6Mb/s.

In these simulations, the wireless nodes were arranged with an AP in the network, which formed a connection be-tween every node in the wireless domain and a single node in the wired domain (see Fig. 1). This AP was connected to the wired network by a high capacity link with negligible delay, which was dimensioned such that it could easily carry all the traffic and hence no loss occurred on this link.

Figure 1: Network Topology

The simulations were configured such that each node used bidirectional traffic sources, so as to model VoIP traffic. This was represented as Constant Bit-Rate (CBR) traffic, transmitted using User Datagram Protocol (UDP). These sources were parameterised to model G.711 voice at 64kb/s with 20ms voice payload. The G.711 scheme was chosen as it is still commonly used, due to its simplicity, despite

the availability of schemes with better compression. A VoIP data payload size of 160 bytes was generated every 20ms, to which the 20 byte IP header, the 12 byte Real-time

Trans-port Protocol (RTP) header and the 8 byte UDP header were

added. This resulted in 80kb/s of traffic being sent in each direction between each node and the AP. This VoIP traf-fic was always transmitted using the highest priority AC of 802.11e in accordance with the standard.

There are two more important delay-related points which must be made. Firstly, due to the delay bounded nature of VoIP traffic, packets which experience excessively high de-lays, will be dropped at the receiver as they will arrive too late to be of use. The ITU-T recommends in the G.114 stan-dard that end-to-end delay should be kept below 150ms, so as to maintain acceptable call quality [7]. Hence, throughout this work 150ms was considered to be the maximum accept-able delay for unidirectional VoIP. The second point relates to internet delay. In early 1998, Kostas et al. [8] chose worst case delays of 100ms. Given the technology improvements which have taken place, a 70ms internet delay budget value has been chosen here.

The above points can be used to arrive at a delay budget for the WLAN. Of the 150ms total delay budget, 70ms can be allocated to internet delay and 30ms is required for the voice encoding, decoding and packetisation processes. The leaves 50ms for access delays. Assuming WLAN connectiv-ity on both ends of the connection, a delay budget of 25ms can be allocated to each WLAN.

4.

OVERVIEW OF PROPOSED SCHEME

The proposed scheme consists of two main parts, a CW adaptation and a TXOP adaptation, with both of these pa-rameters being adjusted every 250 packets. The scheme is quite centralised in that appropriate values for the TXOP and CW are determined at the AP and communicated to each of the nodes in the system.

The aim of the CW adaptation in the proposed scheme is to determine an appropriate CW size which strikes the right balance between loss associated with small CW sizes and delay associated with larger CWs. Using this approach, collisions and retransmissions can be kept to a level which is appropriate for the traffic on the system.

The proposed scheme differs from the standard scheme in that the wireless nodes do not adapt the CW themselves. Specifically, they do not increase their CW range if they ex-perience an unsuccessful transmission. Rather, they main-tain a fixed value for the CW, which is called dCW for

clar-ity, until the AP advises them to update their window size. If a transmission is unsuccessful, a random backoff time is again selected from the range [1, dCW + 1]. If there are many

retransmissions, however, the AP will rather quickly deter-mine a new value for dCW which is appropriate for the

net-work conditions.

Another component of the scheme is the adaptation of the TXOP parameter. Adapting the TXOP parameter in an appropriate way can result in good control over the divi-sion of the available resources between uplink and downlink traffic - something that is important for symmetric VoIP traffic. The specific approach that was used in this work was to vary the TXOP used by the AP and to keep that of the wireless nodes fixed. This approach is reasonable as the uplink/downlink control is more sensitive to the ratio

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between the AP TXOP and that of the wireless nodes than the specific values that are used.

The following section explains the development of the scheme in greater detail. This is followed by a discussion of the choice of appropriate parameters for the scheme and how they were selected.

4.1

Detailed Description of Proposed Scheme

There are three important aspects to the scheme - deter-mining the level of retransmissions on the system, defining how dCW is adjusted and defining how the TXOP

parame-ter is adjusted. Each of these is discussed in the subsections below.

4.1.1

Determining the Level of Retransmissions in

the System

An Exponentially Weighted Moving Average (EWMA) es-timator, as defined in equation 1, was used to determine the level of retransmissions in the system. y(k) is the moving av-erage estimator of the level of retransmissions on the system and x(k) is the number of transmissions packet k has under-gone before it was successfully received at the AP. α is the parameter that determine how much memory the EWMA estimator has. In this case a value of 0.98 was used: this choice of α means that the weight of those samples which are not in the previous 250 samples is minimal.

y(k) = [α ∗ y(k − 1)] + [(1 − α) ∗ x(k)] (1) In order to estimate the retransmission level on the sys-tem, it was necessary to know the number of attempted transmissions required by each packet before it was success-fully received. Here, it was assumed that this information can be obtained from each packet. A small change to the standard can facilitate this. More specifically, an extra field can be added to the MAC header in which the sender inserts the transmission attempt number of each packet.

4.1.2

CW Adaptation

The CW was adapted as follows. After every 250 pack-ets, the AP compared the estimated level of retransmissions with a reference parameter. If the estimate fell below the reference parameter, the CW was decreased and if it was in excess of the reference parameter, the CW was increased.

The following approach, then, was used to adjust the value of dCW .

if (y(k) > γ) then ( dCW (k) = ( dCW (k − 1) + κinc))

where γ is the reference parameter and κincis the

param-eter that controls how much dCW is incremented by.

Sim-ilarly, dCW is decremented if the estimate of the level of

retransmissions is greater than γ using the rule if(( dCW (k) > 7) && (y(k) < γ))

then ( dCW (k) = ( dCW (k − 1) − κdec))

Note that there is a lower bound of 7 on the value of dCW

as is the case in the standard ERP-OFDM scheme.

Appropriate values for the increment and decrement pa-rameters, κinc and κdec were unclear and some simulations

were performed to arrive at suitable values. These are de-scribed below.

4.1.3

TXOP Adaptation

In this scenario, given the use of the G.711 codec and a data payload of 20ms, the time taken to transmit a single packet was approximately 150µs. This value takes into ac-count the delay for data transmission, a SIFS delay, the de-lay for the acknowledgment and the dede-lay for the OFDM sig-nal extension. The time required for the data transmission consists of time spent transmitting the data payload, MAC header, RTP header, UDP header, IP header and PLCP header and preamble. So as to accommodate multiple such transmissions with ease, TXOP adjustments were made in multiples of 155µs.

As all connections go through the AP, it is clear that the AP requires more access to the medium than the other nodes in the system. By assigning a larger TXOP value to the AP, it can transmit more packets once it gains medium access, resulting in greater use of the medium by the AP.

In order to determine an appropriate TXOP size for the AP, counts of both the ACKs successfully received from the wireless nodes and the packets successfully received at the AP were. After every 250 packets received at the AP, these two counts were compared. If more data packets were re-ceived than ACKs, it was an indication that the uplink was performing better than the downlink. At such times the maximum TXOP for the AP was increased by 155µs to en-able the AP to transmit more in the downlink. Conversely, if more ACKs were received than packets, then the maximum TXOP for the AP was decreased by 155µs. No upper bound was put on the TXOP value used by the AP. In this manner, the system was able to adapt to changing traffic levels in or-der to maintain similar uplink and downlink performance levels.

Many simulations were carried out in order to determine a reasonable value for the maximum TXOP used by the wire-less nodes. Results showed that a maximum TXOP value of 1085µs demonstrated improved performance. This repre-sents a maximum of approximately 7 packets being sent in each burst by the wireless nodes. This value was chosen for the experiments below.

4.2

Determination of Appropriate System

Pa-rameters

Two important sets of system parameters must be defined for the above approach to operate. Firstly, the reference pa-rameter, γ must be defined. The meaning of this parameter and a description of experiments performed to arrive at a suitable value for this parameter is given in the subsection below. The other parameters that need to be defined are the increment and decrement parameters, κincand κdec.

Exper-iments were performed to determine appropriate values for these parameters - these are described in the following sub-section.

4.2.1

Determination of

γ

Several series of simulations were performed in order to determine an appropriate level for γ. For these initial ex-periments κinc and κdec were set to 1. In this instance the

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operat-ing point for the system in terms of the fraction of retrans-missions on the system.

Different values of γ were compared on the basis of good-put, loss and delay. γ values of 2%, 5%, 10%, 15%, 20% and 25% were examined. The scheme which provided the highest goodput and call capacity whilst maintaining delay and loss within recommended boundaries was chosen.

Results for the different γ values showed that the 2% and 5% levels exhibited significantly lower goodput than the other retransmission levels and for this reason they were ruled out as possible operating points.

65 70 75 80 85 90 95 100 47 48 49 50 51 52 53 % Packets

Bidirectional VoIP Calls

10% 15% 20% 25%

Figure 2: Percentage of Packets Within 25ms Uplink Delay Budget

The other potential γ levels were then compared (see Fig. 2) on the basis of the call capacity each scheme could handle before packet delays exceeded a 25ms uplink delay budget. From these results it can be seen that both the 10% and the 15% levels began to show increased delay prior to the other schemes. This occurs as these levels of retransmis-sions can only be realised if the values of dCW are large,

re-sulting in large backoff delays and ultimately, unacceptably large delays. Of the 20% and 25% retransmission rates, both schemes seemed to give reasonably good performance. As there was little difference between these two target values in terms of both goodput and delay, either would be equally suitable; the 20% value was chosen.

4.2.2

Determination of

κinc

and

κdec

Experiments using different values for κinc and κdecwere

performed for γ = 20%. The results were compared on the basis of both loss and delay. A scheme which reacted to increases in retransmission levels with large increases in dCW

and then single step decreases, seemed to perform the best. A scheme which incremented in steps of 7 and decremented in steps of 1 was ultimately chosen.

It is worth noting that this choice of κincand κdecmeant

that the system no longer oscillated around γ, i.e. γ was no longer a target operating point which the system continually attempted to converge to. Further, the levels of retransmis-sions on the system were always lower than γ when this approach was used, as the system had a tendency to use a larger dCW . As this choice of κincand κdecresulted in better

performance, they were selected for the experiments below, even though the interpretation of γ was less clear.

4.3

Results and Discussion

In order to assess the performance of the adaptive scheme, it was compared with the original scheme. However, for the traffic patterns under consideration here, the small

maxi-mum CW value from the standard is inappropriate. There-fore, a series of simulations were performed with the maxi-mum CW value of the standard scheme increased to differ-ent multiples of its original value. Thus, the maximum CW values chosen for analysis were 63, 127, 255 and 511.

4.3.1

Goodput

All five scenarios are found to give quite a similar good-put level (see Fig. 3). Although the adaptive scheme per-forms slightly better than the other schemes. However, the difference is very marginal, especially in comparison with the schemes using the larger maximum CW values. The schemes with smaller maximum CW values are seen to per-form slightly worse, which indicates that the maximum CW value in these cases is bounded at a value that results in significant collisions on the medium.

5.8 6 6.2 6.4 6.6 6.8 7 7.2 7.4 46 48 50 52 54 56 58 Mb/s

Bidirectional VoIP Calls

Adaptive 1085us CWmax=63 CWmax=127 CWmax=255 CWmax=511

Figure 3: Comparison of Adaptive Goodput with Original

4.3.2

Loss

ETSI studies [5] indicate that a packet loss rate of 5% is at the quality threshold of fair quality voice. The results in Fig. 4 show that all of the schemes result in acceptable uplink loss. However, there is a difference between the loss levels obtained under the different schemes. The adaptive scheme clearly performs better than the other schemes with negligible loss for the call levels studied.

0 1 2 3 4 5 6 46 48 50 52 54 56 58 Loss Rate

Bidirectional VoIP Calls

Adaptive 1085us CWmax=63 CWmax=127 CWmax=255 CWmax=511

Figure 4: Uplink Loss Comparison

In addition, the downlink loss in all the schemes (see Fig. 5), was found to be low in general. Downlink loss rates did not even reach 2.5% in any of the scenarios studied. Although the adaptive scheme demonstrated the highest downlink loss rates at times, the aggregate loss rate (up-link+downlink) for the adaptive scheme is still significantly lower than that of the other schemes. Also, it is well below the loss threshold of 5%.

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0 1 2 3 4 5 6 46 48 50 52 54 56 58 Loss Rate

Bidirectional VoIP Calls

Adaptive 1085us CWmax=63 CWmax=127 CWmax=255 CWmax=511

Figure 5: Downlink Loss Comparison

4.3.3

Delay

The percentage of packets that arrive within a delay bud-get of 25ms are now compared for all five schemes. It was found that the number of packets received within the delay budget on the downlink remain acceptable, at more than 99%, in all five cases for the call levels studied. In contrast, delays on the uplink (see Fig. 6) rise sharply at or before 49 calls in the schemes using maximum CW values of 63 and 127. This is as a result of the traffic levels on the sys-tem, as in this situation the higher maximum CW values cause a decrease in the level of retransmissions. This results in a higher percentage of packets being received within the specified delay budget.

0 20 40 60 80 100 46 48 50 52 54 56 58 % Pkts

Bidirectional VoIP Calls

Adaptive 1085us CWmax=63 CWmax=127 CWmax=255 CWmax=511

Figure 6: Uplink Packets Within Delay Budget of 25ms

At 48 calls the 255 CW scheme already shows 4.04% of packets exceeding the uplink delay budget and the 511 CW scheme shows 3.76% of packets exceeding this budget. How-ever, the adaptive scheme shows that less than 3% of packets exceed the uplink budget until after the 52 call point, thus showing its superior ability to adjust to network conditions. This shows that an additional 4 calls can be accommodated when using the adaptive scheme according to this uplink delay budget.

When higher uplink delay budgets were used, similar re-sults were obtained. Rere-sults show that delays in the other schemes reach higher levels much more quickly, whereas the adaptive scheme demonstrates a more gradual increase in both loss and delays, something which is quite desirable in WLAN systems.

5.

CONCLUSION & FUTURE WORK

A mechanism for jointly adapting the contention window and TXOP parameter on the basis of successful and

unsuc-cessful transmissions is outlined. The mechanism dynami-cally adapts these MAC parameters so as to maintain ac-ceptable performance levels for higher call capacities. The dynamic nature of this scheme takes into account the cur-rent conditions of the WLAN and adapts the parameters accordingly.

Comparisons using a series of simulations using the stan-dard 802.11e/802.11g scheme but using different maximum CW values were discussed in detail. Results show that the adaptive scheme dynamically adjusts the contention window and TXOP to give similar goodput to variants of the stan-dard scheme with increased CW but with lower delays and loss due to the effect of both the TXOP and the dCW being

adjusted in parallel. Thus, the adaptive scheme can consis-tently provide a higher VoIP call capacity, for the studied scenarios.

Future work involves examining the operation of the pro-posed scheme in conjunction with the other QoS classes out-lined in the IEEE 802.11e MAC. An examination of the scheme operating at the different data rates available to IEEE 802.11g, as well as with a range of VoIP codecs, is also to be performed. Finally, a modified version of the scheme is planned, which dynamically adapts on the basis of goodput levels rather than retransmission rates.

Acknowledgements

The support of the Informatics Research initiative of Enter-prise Ireland is gratefully acknowledged.

6.

REFERENCES

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Distributed Coordination Function”, IEEE Journal on Selected Areas in Communications, 18(3):535-547, Mar. 2000

[3] G. Bianchi, “IEEE 802.11-Saturation Throughput Analysis”, IEEE Communications Letters, 2(12), Dec. 1998

[4] C. Casetti and C.F. Chiasserini, “Improving Fairness And Throughput For Voice Traffic In 802.11e EDCA”, Proc. of IEEE PIMRC’04, Barcelona, Spain, Sep. 2004

[5] ETSI TR 101 329-6 V2.1.1 (2002-02) Technical Report Telecommunications and Internet Protocol Harmonization Over Networks (TIPHON) Release 3; Part 6

[6] D. Gu and J. Zhang, “A New Measurement-Based Admission Control Method for IEEE 802.11 Wireless Local Area Networks”, Proc. of IEEE PIMRC, pp. 2009-2013, Sep. 2003 [7] ITU-T Recommendation G.114: One Way Transmission Time,

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[8] T. J. Kostas, M. S. Borella, I. Sidhu, G. M. Schuster, J. Grabiec and J. Mahler, “Real-Time Voice Over

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[9] S. Choi and J.del Prado Pavon, “802.11g CP: A Solution for IEEE 802.11g and 802.11b Inter-Working”, Proc. of VTC-Spring’03, Jeju, Korea, Apr. 2003.

[10] D. Pong and T. Moors, “Using Transmission Opportunities and Judicious Parameter Selection in Enhancing Real-Time Applications over 802.11 Wireless LANs”, Proc. of ATNAC’03, Dec. 2003

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