A New Admission Control Metric For Voip Traffic in 802.11 Networks
A New Admission Control Metric For Voip Traffic in 802.11 Networks
11 Networks
Sachin Garg sgarg@avaya.com Avaya Labs Research 233 Mount Airy Rd. Basking Ridge, NJ 07920 Martin Kappes mkappes@avaya.com Avaya Labs Research 233 Mount Airy Rd. Basking Ridge, NJ 07920
Abstract In this paper, we propose a new metric for admission control of VoIP trafc for IEEE 802.11b networks. Due to the large xed overhead in IEEE 802.11b, the bandwidth available at the payload sizes typical for VoIP trafc is far less than the bandwidth available when using the network for data trafc. Hence, bandwidth is inappropriate as criterion whether or not a new VoIP stream or data stream can be accommodated without pushing the network over its throughput limit. We propose a new criterion called network utilization characteristic which allows to assess the available resources in the wireless network in an accurate way and show how it can be used for admission control.
1 Introduction
In the last few years, wireless networks based on the IEEE 802.11b standard have gained popularity and have been widely deployed in enterprises mostly to provide wireless data access from Laptops, PDAs, etc. to the wired infrastructure of the enterprise. They have also been deployed in public hot-spots such as airports, hotels, conference facilities etc., mainly for internet connectivity. The maximal data rate 802.11b currently supports is 11 Mbps (other possible data rates are 1, 2 and 5.5 Mbps). When sending data frames with this rate, the maximal throughput achievable in such a network is approximately 6.2 Mbps. For VoIP trafc, the maximal throughput achievable is approximately only 2 Mbps for typical audio payload size per RTP packet. The signicant difference is due to the large transmission overhead per frame which remains the same regardless of the frame size. Depending on the actual average transmission rate, the number of simultaneous VoIP calls in a cell 1 of the wireless network is between 4 and 17 with G711 codec with 30ms audio data per packet. For details, see [6].
1
we use the term cell for what is referred to in the 802.11 standard as Basic Service Set (BSS).
As converged networking in the wired world gains foothold, it is likely that wireless networks will also be increasingly used for voice trafc. Placing an additional call or an additional data connection that exceeds the capacity of the wireless network will likely result in unacceptable call quality for all ongoing VoIP calls. Further, if the load offered to the network is higher than its capacity, the DCF medium access scheme of 802.11 curtails the client with the highest load rst. In most cases, the access point of the wireless cell puts more trafc on the air than the associated stations. Hence, it gets curtailed rst which leads to unacceptable packet loss for all VoIP streams transmitted from the access point to a client resulting in bad call quality for all connections. Thus, taking into account the low number of VoIP connections possible, the need for VoIP admission control is apparent. Experimental results in [6] show that the average delay incurred on the wireless link for VoIP connections is about 5ms and the average jitter is about 7-9ms even when the network load is low. These values are signicantly higher than in a wired environment, for instance, Ethernet. Hence, MAC layer improvements to lower the delay and jitter values, such as the proposal from the 802.11e taskgroup, are very important. Yet, any QoS strategy for networks, where the available network capacity is limited and likely to be exceeded with consequences for all connections needs to provide an accurate way of measuring the network capacity and of enforcing admission control. In fact, the methods for measuring network capacity and providing admission control outlined here could blend in well with 802.11e.
Denitions
2.1 Flows
The trafc on the wireless network is partitioned into ows. Each transmitted frame on the wireless network belongs to exactly one ow. The criterion could be frame/packet identiers at various network layers starting from the MAC Layer (Layer 2) up to the Transport Layer (Layer 4). A frame is classied based on a lower layer criteria only if the higher layer information is unavailable of the packet/frame belongs to a specic layer. For instance, ICMP packets are Layer-3 packets. Further, certain frames such as Beacons, probe-request and response frames are limited to Layer-2 only. In general, using higher layer criteria, if possible, provides ner granularity of classication. A Transport Layer ow is uniquely determined by the 4-tuple
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and by the Transport Layer protocol type of the ow, namely TCP or UDP. Given two communicating stations, this classication scheme implies that the network utilization is measured separately for each trafc direction. A TCP connection results in two TCP ows and a VoIP call consists of two UDP 2
ows. In other words, this separation captures the assymetric nature of most data connections, such as le/web downloads in terms of network utilization. Apart from ows associated with voice and data trafc, we also consider auxiliary ows. Auxiliary ows represent network activities such as erroneous transmissions and collisions which cannot be accredited to any particular ow but typically represent wasted network capacity.
that can be transmitted per second is 1193, resulting in a maximal throughput of 954 Kbps. For 1000 bytes size per frame, a maximum of 670 frames can be transmitted per second, resulting in a maximal throughput of 5.36 Mbps. Hence, the simple questions can the network handle a ow with 2 Mbps bandwidth or can the network accommodate 800 packets per second cannot be answered without additional information. The situation becomes even more complex when multiple ows are to be considered as necessary for admission control. Therefore, we propose a measure called network utilization characteristic, which is measured on a per-ow basis. The network utilization characteristic (NUC) of a ow is dened as the fraction of time per time unit needed to transmit the ow over the network. Two ows having the same network utilization characteristic are said to be bandwidth equivalent. In the remainder of this paper, we will base our considerations on a per second basis. The choice is arbitrary. Any other interval, such as a beacon-period can be used without any change in the results. First, we illustrate this concept by some examples. Consider a ow with 100 byte size frames having a bandwidth of 100 Kbps and transmission parameters as outlined above. Then the time (overhead and actual data transmission time) to transmit a single frame is 837 s. Transmitting 100 Kbps using a frame
0.9
0.8
0.7
0.6 NUC
0.5
0.4
0.3
0.2
0.1
0
1 51 10 1 15 1 20 1 25 1 30 1 35 1 40 1 45 1 50 1 55 1 60 1 65 1 70 1 75 1 80 1 85 1 90 1 95 1 10 01 10 51 11 01 11 51 12 01 12 51 13 01 13 51 14 01 14 51
Figure 1: NUC as a function of packet-size for xed bandwidths in the scenario of a single client sending at 11 Mbps. size of 100 bytes requires 125 packets per second. Transmitting 125 packets of that size takes 104.6 ms. Hence, the NUC of the ow is 0.1046. Now consider a ow with 1000 byte size frames having a bandwidth of 1 Mbps and the transmission parameters as outlined above. The ow sends 125 frames per second and the transmission time for a single frame is 1492 s. Hence, the NUC of the ow is 0.1865. Figure 1 shows the NUC as a function of packet-size for some xed values for the bandwidth for a single client sending at 11 Mbps. As can be seen, the NUC of a ow can range almost anywhere from 0 to 1 for a given xed bandwidth. This gure stresses the necessity for using NUC instead of bandwidth for assessing the capacity of a wireless network. Summing up the NUCs of all ows (including auxiliary ows) in the network yields the fraction of time the network is busy. Consequently, the difference between one and the sum is the time the medium is idle. A ow can be accommodated without sacricing other ows if its NUC is going to be smaller than this value.
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SIFS
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Figure 2: IEEE 802.11 CSMA/CA medium access scheme. that ow per second and the average transmission time for a frame. It is straigthforward to monitor packets per second, used bandwidth and average packet size per ow as outlined in [5]. For regular Ethernet, these three parameters would be sufcient to compute the transmission time per packet. For 802.11 networks, however, the overhead due to the channel access mechanism is not captured in any of the parameters. In fact, this overhead is substantial and cannot be ignored. We now briey describe the Collision Avoidance (CSMA/CA) medium access scheme according to the Distributed Coordination Function (DCF) of the 802.11 standard [7] in order to explain the associated overhead. The MAC protocol is designed to prevent collisions from occurring. Furthermore, unicast frames are acknowledged by the receiving station. The acknowledgment (ACK) is sent out after the transmission has nished and a certain duration of time called short inter frame spacing (SIFS) has elapsed. If a node wants to transmit a frame and senses the medium idle for a certain duration of time called distributed coordination function inter frame spacing (DIFS), it may start transmitting. As DIFS is longer than SIFS, it is made sure that a correctly received frame can always be acknowledged before the next frame is transmitted. If a node wants to start transmitting while the medium is busy or if it wants to transmit another frame after just nishing a transmission, it also waits for the medium to be idle for the DIFS period. Then, the node does not begin to transmit immediately but enters a contention phase for the medium. Contention is done by choosing an integer random back-off from a certain interval. The random backoff determines the number of time slots the client defers its transmission in addition to the DIFS time. If the medium is sensed idle in such a slot, the back-off timer is decreased by one. If the random back-off has decreased to , the node starts transmitting. If another node starts transmitting before this happens, the node continues to count down the back-off timer after the medium has been sensed idle for the DIFS period. Thus, if multiple clients want to transmit a frame, the one with the lowest random back-off time will win the contention for the medium.
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SIFS DIFS ACK
time
the size of the data frame in bytes. Then, the time needed
for transmitting the frame can be computed as shown in Table 1. Apart from frame size, the only information needed to calculate the tabulated values is the transmission speed. The component of the transmission time not addressed so far is the back-off value. As our perspective is the usage of the wireless medium, we are not interested in the actual back-off window that was chosen for the transmitted frame but in the actual number of idle back-off slots immediately preceding the transmission. A slot time in 802.11b is 20 s, so the number of slots waited between the end of DIFS and the transmission multiplied by the slot time yields the desired value.
can be computed as
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Note that the NUC obtained is identical with the number which would be computed by summing up the transmission time needed for all frames of the ow. Whereas the number of frames sent per second and the average number of bytes sent per data frame can be obtained by standard means (such as SNMP MIBs), the transmission speed can be observed by both the receiving and the sending party. As the admission control scheme is most likely to be implemented in the access point and as all trafc in an infrastructure-based network either origins from the AP or is destined to it, it is feasible to get accurate information about the actual transmission speed of a particular frame and thus also of the average transmission speed of a particular ow. The transmission speed is determined by the wireless station based on factors such as frame error rates, strength of the radio signal etc. and is not dependent on the characteristics of a ow. In fact, all ows emnating from a station will use the same transmission speed at a given time. Therefore, it sufces to measure this parameter on a per-station basis and use the value for all ows from that station. Moreover, instead of recording the transmission speed of every frame, it may be sufcient to compute the average transmission speed based on sampling frames in a short period such as a beacon-period. Determining tha actual back-off before transmission on a per-frame basis is not possible for anyone but the station transmitting the frame. An observer on the channel cannot distinguish between idle times caused on the channel because of back-off slots before a transmission versus the idle time caused because the station did not attempt to transmit at all. However, for our purposes, we are interested in an average back-off value on a per-second basis. Due to the fair nature of DCF, the average back-off experienced by any station is the same. In other words, the average back-off can be measured at the access-point and the same value used for all stations. Like before, it may sufce to measure average back-off based on few random samples at the access-point in a xed duration. As the preceding discussion shows, the values necessary to determine the NUC of a ow can be easily derived by standard means or accurately estimated even without having full access to the PHY/MAC layers of the AP. In fact, we believe that there is a spectrum of possibilities trading off accuracy against simplicity of data collection. For instance, in order to assess the aggregated NUC of all ows in the network, it is sufcient to know the number of packets sent, the bytes sent, the average transmission speed in the network and the average actual back-off. While we only addressed how to compute or estimate the NUC of ows, it is also apparent that the data needed to compute the NUC of auxiliary ows, for instance due to collisions or erroneous transmissions, is present in the AP. For other ways of accounting for such trafc, see Section 4.1.
ow cannot create signicant load on the network before it is detected. VoIP ows consist of real-time audio data. Such ows typically use connection-less transport layer protocols, in particular UDP, that do not guarantee the delivery of sent audio data information. In other words, lost packets are not retransmitted. Furthermore, the amount of trafc that the sender is transmitting to the receiver is xed for the duration of the ow. Thus, it will not be inuenced by parameters such as current load conditions of the network as indicated by e.g., lost packets or ICMP messages. Although losing some packets once in a while does not render the ow useless, the loss of more than 1% of the data usually deteriorates the quality of the ow to an unacceptable level. This in turn implies that curtailing such a ow by dropping a certain percentage of the packets renders the whole ow useless. Therefore, such a ow must either be given the full bandwidth it requires or be completely dropped. The bandwidth such a ow needs can be easily detected either by transparently examining the messages exchanged between the endpoints of the ow during connection setup time or by just observing the trafc patterns. As the average transmission speed of all clients in the wireless network and the average actual back-off are known, the NUC for that ow can be accurately estimated. It should be noted that while the number of frames and the size of the sent frames remains constant during the lifetime of the ow, the NUC of such a ow can change signicantly due to variations in the transmission data rate, especially in the case of mobile clients. We will discuss how to deal with this situation in the following section. Data ows, on the other hand, use connection-oriented transport layer protocols such as TCP. In most cases such protocols also provide ow control and congestion control. Apart from resending lost data, ow control and congestion control adapt the network usage of the connection due to current conditions such as available buffer size on the receiving side or network congestion. In other words, the bandwidth used by such a ow as well as its NUC can change over time. Whereas the user may notice a smaller bandwidth e.g., by longer transmission time for a le or web page, a reduction in bandwidth does not render the ow useless. Therefore, such ows can be throttled down by means such as TCP congestion control. The use of TCP congestion control and other mechanisms to curtail the bandwidth of such a ow is described in [5]. As opposed to VoIP ows, the bandwidth of a data ow cannot be assessed from the values collected when the ow is detected. In what follows, we assume that no information about the NUC of the ow is available. Table 2 summarizes the difference between VoIP and data ows. The goal of this paper is to show how NUCs can be used for admission control for VoIP in wireless networks. The actual policies however are not subject of this paper. Therefore, we will present a
VoIP Flow Can be accurately determined when ow is detected Do not change during the lifetime of the ow Flow must either be given full NUC needed or must be shut down
Data Flow Cannot be determined when ow is detected Do change during the lifetime of the ow Flow can be curtailed (almost) arbitrarily. Flow restrictions are changed over time.
Table 2: Comparison between VoIP and data ows. somewhat more abstract presentation of how the permissibility of a ow is computed. The steps for a newly detected VoIP-ow are as follows. NUCTotal is computed and the NUC for the new VoIP-ow are estimated. If NUCTotal plus the NUC for the new ow is less than NUCTotalMax, the VoIP ow is admitted. If not, determining on the policies a decision whether other ows are to be curtailed for the new ow or not is made. If so, the restrictions are calculated and enforced and the new ow is admitted. If not, the new ow is not admitted. Clearly, calculating the NUCs as described above generates accurate data of the usage of the network in a past interval. Our assumption is that the past NUC of a ow constitutes a good estimate of the ows future NUC. In other words, we assume a steady state usage model for computing the permissibility of a new VoIP ow. As in fact the bandwidth used by a ow may change over time, it is necessary to also enforce bandwidth restrictions of non-VoIP ows in order to provide VoIP admission control. This holds especially true if the network operates close to its capacity limits. Along the same lines, when a new data ow is detected, the NUC available for this ow is to be determined and enforced depending on the policies in the system. While the NUC of a ow accurately measures the network resources the ow uses, we think that bandwidth is a variable that will probably also be taken into used in formulating policies for admission control. Apart from determining the question whether new ows should be admitted or not, the NUCs of all ows need to be monitored constantly.
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certain inaccuracies that might arise if data needed to determine the NUC of ows and auxiliary ows is only partially collected on one hand and the use for creating a network capacity backup that might be needed in some situations on the other.
to
. With reasonable safety we may assume that actual collisions will not exceed that value and thus
that if NUCTotal is less than NUCTotalMax all ows can be transmitted without loss or curtailment due to exceeding the network capacity. Whereas this approach is elegant and simple, in most cases the NUC wasted by collisions will be less than 15 % and hence some capacity of the network would be wasted. Therefore, a more accurate estimation of the collisions would be benecial. As extensive simulation studies have shown [1] [2] [3] [4], the number of collisions is a function of the number of clients associated with an AP and their trafc characteristics. Hence, we could tabulate those values and then estimate the NUC of the wasted channel capacity by looking up those values. This would allow for a more precise treatment of collisions. In fact, one could think of a large variety of schemes and situations how to use NUCTotalMax along the lines indicated here.
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In a network mostly used for data connections, this increase in the NUC of a VoIP ow can be accommodated by further curtailing the NUC of data connections. However, problems can occur if the network is prevailingly used for VoIP. As no ow can be curtailed then, there is the need to have some network capacity reserve that could accommodate a change in transmission speed for some of the connections. As it is unlikely that all stations roam out of range at the same time, this reserve would probably be large enough if it consisted of sufcient NUCs for one or two slow bandwidth connections. However, whether such reserves need to be present or not is a question of policy.
5 Prototype Implementation
In the future, we are planning to implement admission control for VoIP connections in wireless networks by employing our Wireless Access Server (WAS) infrastructure as described in [5]. Figure 3 shows the high-level setup of the system and how it is deployed in a typical enterprise network. Alternate congurations to this setup are possible. WAS consists of two components. The rst component is a box, which sits between the wired network and the wireless-network. Specically, the box sits at the edge of wired network immediately behind the access-points. All trafc to/from an access-point traverses this box. This box, referred to as the Wireless Gateway (WG), acts as bridge with ltering capabilities at the IP and TCP/UDP layer. In other words, the Wireless Gateway may operate completely transparent to the clients in the wireless network. Multiple access-points could be bridged (connected) via a single gateway. This basic setup provides us with a platform for QoS, access control and other features for 12
Figure 3: High-level System Architecture. 802.11 networks. Note that the functionality proposed for WG need not be present in a separate box. In fact, we are planning to move the WAS functionality into an access point running embedded Linux. The second component of Wireless Access Server is called the Gateway Controller (GC). The GC may reside on any point of the wired network. The GC is responsible for controlling the behavior of the wireless gateway.
6.1 RTS/CTS
Apart from the standard DCF scheme in IEEE 802.11 standard [7] as outlined before, it also provides a Ready to Send / Clear to Send (RTS/CTS) extension that is particularly useful in wireless networks which might suffer from the hidden station problem. The channel usage of a single frame transmission with RTS/CTS is shown in Figure 4. As the standard species, the use of the RTS/CTS mechanism is specied on a per-station basis and each station can be congured to use RTS/CTS either always, never or only on frames longer than a specied length. Hence, if the policy of the sending station and the length of the sent frame is known, it can be determined whether RTS/CTS is used or not even without observing the transmission. The policy of the each station in the network can either be obtained by observing its behavior or by querying it from the station. 13
BACKOFF RTS
SIFS
SIFS
CTS
Figure 4: IEEE 802.11 CSMA/CA medium access scheme with RTS/CTS On a per-ow basis, the use of RTS/CTS can be estimated by the average size of frames belonging to the ow and the variation of it. As shown in Figure 4, the overhead added by RTS/CTS consists of two additional SIFSs and the overhead to transmit the 20 byte RTS and the 14 byte CTS frames. Similar to the values shown in
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respectively and are thus depending on the transmission rate. We treated the case of RTS/CTS in some detail as an example how the NUCs of ows can be accurately computed in other scenarios. In fact, this can be done for other special cases as well as for instance if fragmentation occurs. In contrast to Ethernet, fragmentation in wireless networks is in most cases not the result of a lower maximal frame size of the wireless link (the maximal frame size in 802.11 networks is far higher than in wired Ethernet) but in most cases done deliberately for improving interference stability of the wireless network. The use of fragmentation results in an overhead that is very similar to the one for RTS/CTS.
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opportunities but does not provide any means to determine the feasibility of such a request.
References
[1] G. Anastasi and L. Lenzini. QoS provided by the IEEE 802.11 wireless LAN to advanced data applications: a simulation study, In Wireless Networks, Vol. 6, pp 99108, J.C. Baltzer AG, Science Publishers, 2000. [2] G. Bianchi. Performance Evaluation of the IEEE 802.11 Distributed Coordination Function. In IEEE Journal on Selected Areas in Communication, Vol. 18, No. 3, March 2000, pp. 535547. [3] A. Heindl and R. German. Performance modeling of IEEE 802.11 wireless LANs with stochastic Petri nets. Performance Evaluation, 44 (2001), 139-164. [4] H. S. Chayya and S. Gupta. Performance modeling of asynchronous data transfer methods of IEEE 802.11 MAC protocol. In Wireless Networks Vol 3, 1997, pp. 217-234. [5] S. Garg, M. Kappes and M. Mani. Wireless Access Server for Quality of Service and Location Based Access Control in 802.11 Networks, submitted. [6] S. Garg and M. Kappes. On the Throughput of 802.11 Networks for VoIP, submitted. 15
[7] IEEE
802.11,
11a,
11b
standard
for
wireless
Local
Area
Networks.
http://standards.ieee.org/getieee802/802.11.html
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