Computer Networks 54 (2010) 3–12

Contents lists available at ScienceDirect

Computer Networks

journal homepage: www.elsevier .com/ locate/comnet

Admission control scheme based on priority access for wireless LANs q

Sunmyeng Kim a,*, Young-Jong Cho b, Yong K. Kim c

a School of Computer and Software Engineering, Kumoh National Institute of Technology, Gumi, Republic of Koreab Department of Information and Computer Engineering, Ajou University, Suwon, Republic of Koreac School of Electrical Electronics and Information Engineering, Wonkwang University, Iksan, Republic of Korea

a r t i c l e i n f o

Article history:Received 24 November 2008Received in revised form 7 April 2009Accepted 9 July 2009Available online 16 July 2009

Responsible Editor: E. Gregori

Keywords:EDCAIEEE 802.11eQoSAdmission controlPriority access

1389-1286/$ - see front matter � 2009 Elsevier B.Vdoi:10.1016/j.comnet.2009.07.002

q This paper was supported by Research FunInstitute of Technology.

* Corresponding author. Tel.: +82 54 478 7547; faE-mail addresses: sunmyeng@kumoh.ac.kr (S. K

(Y.-J. Cho), ykim@wku.ac.kr (Y.K. Kim).

a b s t r a c t

In order to support the quality of service (QoS) requirements at the medium access control(MAC) layer, the enhanced distributed channel access (EDCA) has been developed in IEEE802.11e standard. However, it cannot guarantee the stringent real-time constraints of mul-timedia applications such as voice and video without an efficient method of controllingnetwork loads. In this paper, we propose a measurement-based admission control scheme,which is made up of two parts: priority access and admission control. First, in order to mea-sure the channel status per traffic type, we propose a priority access mechanism in whicheach priority traffic is distinguished by a busy tone, and separately performs its own packettransmission operation. Then, admission control mechanism protects existing flows fromnew ones, and maintains the QoS of the admitted flows based on the measured channelstatus information. Performance of the proposed scheme is evaluated by simulation. Ourresults show that the proposed scheme is very effective in guaranteeing the QoS of multi-media applications as well as in avoiding the performance starvation of low prioritytraffics.

� 2009 Elsevier B.V. All rights reserved.

1. Introduction

The IEEE 802.11 wireless LAN is widely used for wire-less access due to its easy deployment and low cost. TheIEEE 802.11 standard defines a medium access control(MAC) protocol for sharing the channel among stations[1]. The distributed coordination function (DCF) was de-signed for a contention-based channel access. The DCFhas two data transmission methods: the default basic ac-cess and optional RTS/CTS (request-to-send/clear-to-send)access.

The widespread use of multimedia applications requiresnew features such as high bandwidth and small averagedelay in wireless LANs. Unfortunately, the IEEE 802.11

. All rights reserved.

d, Kumoh National

x: +82 54 478 7539.im), yjcho@ajou.ac.kr

MAC protocol cannot support quality of service (QoS)requirements [2,3]. In order to support multimedia appli-cations with tight QoS requirements in the IEEE 802.11MAC protocol, the IEEE 802.11e has been standardized[4]. It introduces a contention-based new channel accessmechanism called enhanced distributed channel access(EDCA). The EDCA supports the QoS by introducing four ac-cess categories (ACs). To differentiate the ACs, the EDCAuses a set of AC specific parameters, which include mini-mum contention window, maximum contention window,and arbitration inter-frame space (AIFS).

In the previous work [2,5–7], the authors focused onstudying the service differentiation. However, service dif-ferentiation alone is not sufficient to provide the QoS ofmultimedia applications without an efficient method ofcontrolling network loads. In this paper, we focus onadmission control mechanism for the IEEE 802.11e EDCA.

Many admission control mechanisms for 802.11e EDCAhave been proposed. The previous mechanisms areclassified into two approaches: model-based and

4 S. Kim et al. / Computer Networks 54 (2010) 3–12

measurement-based admission control [8]. Model-basedadmission control approach uses an analytical model to de-cide whether a new flow is admitted or rejected [9–12]. Mea-surement-based approach uses measurement information(e.g., throughput, collision probability, or delay) of existingtraffic in a network to make admission decisions [12–15].

The previous mechanisms have several problems. First,the analytical models are usually derived based on a fewunpractical hypotheses to calculate the QoS metrics of allexisting flows in addition to a new one. They do not reflectthe characteristics of real traffics. Therefore, the model-based mechanisms are always inaccurate and clearly notapplicable to realistic environments. Second, it is hard formeasurement-based schemes to make a right admissiondecision since the schemes cannot accurately predict thechannel time usage after a new flow is admitted to the net-work. For example, the scheme proposed in [13,14] allo-cates the bandwidth to flows and makes an admissiondecision based on successful transmission time and a con-stant parameter SurplusFactor. The parameter is used tocompensate potential transmission failures because thisscheme does not recognize which AC flows send packetswhen there are collisions. The inaccuracy of this parametercan lead to making a wrong decision. Third, the previousschemes do not solve the priority reversal problem thatlow priority flows have a shorter backoff counter thanhigher priority ones, and consequently access the channel[16]. Since each flow decreases its backoff counter whenthe channel is idle, it will eventually have a small valuealthough its initial counter is large. A high priority flowcannot be always assured to have a smaller backoff countersince the backoff counter is randomly selected based onthe uniform distribution. Therefore, lower priority flowscan transmit their packets prior to higher priority ones.As a result, higher priority traffic may wait a long timefor the channel contention. Fourth, although an admissioncontrol scheme works well, higher priority traffics are vul-nerable to lower priority traffics since the latter ones maycause many collisions with the former ones. Finally, mostof the previous schemes first regulate data traffic in orderto provide the QoS to the real-time traffics. Therefore, theperformance starvation of data traffic can be caused at highloads.

In this paper, we propose a measurement-based admis-sion control scheme, which is made up of two parts: prior-ity access and admission control. First, the priority accessmechanism uses a short duration busy tone signal (i.e.,pulses of energy) [17]. Each flow sends a busy tone in thelast slot of its own AIFS period, and then it operates backoffprocedure like the EDCA. However, flows that receive thebusy tone suspend their busy tone transmission and back-off procedure. This ensures that each priority traffic sepa-rately performs its own packet transmission operation.Second, the admission control mechanism regulates net-work loads and protects admitted flows from new ones.The priority access mechanism is used to facilitate theadmission control. The access point (AP) measures thechannel status information for each traffic type, and thenseparately makes an admission decision.

The proposed scheme makes the following contribu-tions: (1) The priority access mechanism blocks the trans-

missions of lower priority ACs when higher priority onehas packets to transmit. This ensures that flows of thehighest AC always access the channel first. Therefore, theproposed scheme can solve the priority reversal problem.(2) The priority access mechanism also ensures the chan-nel contention to be among flows with the same priorityAC. In other words, there are no channel contentions andcollisions among flows with different priority ACs. There-fore, the AP exactly measures the channel status informa-tion for each priority AC, and then accurately predicts thechannel time usage after a new flow is admitted. (3) Band-width allocation to each priority AC in the previousschemes is extremely challenging due to collisions amongdifferent priority ACs. However, the proposed scheme canallocate the bandwidth to each priority AC exactly sincethere are no collisions among different priority ACs.Accordingly, the proposed scheme can prevent the starva-tion of data traffic.

The paper is organized as follows. In Section 2, we give abrief introduction of the IEEE 802.11e EDCA and thescheme proposed in [13,14]. In Section 3, the proposed pri-ority access mechanism is presented in detail. In Section 4,we describe the admission control mechanism based onthe priority access. In Section 5, performance studies arecarried out through simulation results. Finally, we draw aconclusion in Section 6.

2. Related work

2.1. IEEE 802.11e EDCA

To enhance the QoS support of the IEEE 802.11 WLAN,the IEEE 802.11e has been standardized [4]. It introducesa new medium access method called hybrid coordinationfunction (HCF), which combines a contention-based en-hanced distributed access mechanism (EDCA) and a con-trolled channel access mechanism (HCCA). The EDCA isan enhanced variant of the DCF. In the DCF, all stationscontend for the channel with the same priority. On theother hand, the EDCA supports several priority levels byintroducing an access category (AC) concept. A station haveup to four ACs to support eight user priorities. Each AC isimplemented as a separate queue. Each packet arrives atthe MAC layer with a priority from higher layer, and ismapped to one AC according to the priority. AC 3, AC 2,AC 1, and AC 0 are for voice, video, best effort data, andbackground traffics, respectively. In order to differentiatethe traffics, the EDCA uses a set of AC specific parameters,which include minimum contention window (CWmin[i]),maximum contention window (CWmax[i]), and arbitrationinter-frame space (AIFS[i]) for AC i ði ¼ 0; . . . ;3Þ. The AIFS isat least distributed inter-frame space (DIFS) long and iscalculated with the arbitration inter-frame space number(AIFSN[i]). The duration of AIFS[i] is defined byAIFS½i� ¼ SIFSþ AIFSN½i� � aSlotTime, where SIFS is a short in-ter-frame space and aSlotTime is the duration of a slot time.For 0 6 i < j 6 3, the EDCA has CWmin½i�P CWmin½j�;CWmax½i�P CWmax½j�, and AIFSN½i�P AIFSN½j�. Note thatin the above inequalities, at least one must be ‘‘not equalto”. The EDCA assigns a smaller CWmin,CWmax, or AIFS to

S. Kim et al. / Computer Networks 54 (2010) 3–12 5

higher priority AC in order to provide the higher chance ofaccessing the channel. Therefore, in the EDCA, the supportof QoS can be achieved by differentiating the probability ofaccessing the channel among different priority ACs.

2.2. Distributed admission control scheme

Xiao et al. proposed a distributed admission control(DAC) scheme, which consists of two levels [13,14]. Atthe first level, the admitted voice and video flows are pro-tected from new and other voice and video flows. The APmeasures the amount of channel time (TxTime[i]) occupiedby successful transmissions from each AC i during a beaconperiod. And then, it calculates the additional amount ofchannel time (TXOPBudget[i]) available for AC i during thenext beacon interval.

TXOPBudget½i� ¼ maxðATL½i� � TxTime½i�� SurplusFactor½i�;0Þ; ð1Þ

where ATL[i] is the maximum amount of channel time forAC i per beacon interval, and SurplusFactor[i] is used tocompensate potential transmission failures due to colli-sions. The budget information is delivered to each stationvia a beacon frame.

Each station maintains the following variables for AC i:TxUsed[i], TxSuccess[i], TxLimit[i], TxRemainder[i], andTxMemory[i]. TxUsed[i] is the amount of channel timeoccupied by transmissions, regardless of success or not,from AC i of this station. TxSuccess[i] is the amount of chan-nel time occupied by successful transmissions. A stationdoes not transmit a data packet when TxUsed[i] exceedsTxLimit[i], where how to determine this value is describedbelow. If a station is prevented from sending a frame forthis reason, TxRemainder½i� ¼ TxLimit½i� � TxUsed½i�. Other-wise, TxRemainder½i� ¼ 0. TxMemory[i] memorizes theamount of channel time that AC i of this station utilizedduring a beacon interval, and is TxMemory½i� ¼a � TxMemory½i� þ ð1 � aÞ � ðTxSuccess½i� � SurplusFactor½i�þTXOPBudget½i�Þ. a is a smoothing factor. TxLimit[i] is definedby TxLimit½i� ¼ TxMemory½i� þ TxRemainder½i�.

The admission decision criterion is as follows:

TXOPBudget½i�P / � ReqBudget½i�; ð2Þ

where ReqBudget[i] is the budget required for a new flow ofAC i, and / is a fraction. If this equation is satisfied, the newflow is admitted.

At the second level, the voice and video flows are pro-tected from best-effort data traffic. In order to control datatransmissions according to traffic conditions, the DACscheme dynamically adjusts the EDCA parameters for datatraffic. This scheme increases the contention window sizemuch faster than the standard by using a larger contentionwindow increasing factor ri for the backoff stage i insteadof 2 used in the standard when a collision occurs. That isri < rjðr1 P 2Þ for 1 6 i < j 6 Lretry, where Lretry is the retrylimit. Whenever a data frame reaches the retry limit, CWminand AIFS increase (i.e., CWmin ¼ h � CWminðh > 1Þ andAIFS ¼ w � AIFSðw > 1Þ). Whenever a station successfullytransmits m consecutive packets, CWmin and AIFS decrease(i.e., CWmin ¼ CWmin=hðh > 1Þ and AIFS ¼ AIFS=wðw > 1Þ).

3. Priority access

In this section we describe the priority access mecha-nism that ensures the channel contention-based on trafficpriority. We consider a single hop IEEE 802.11e network.Our mechanism needs minor modifications to the EDCA.However, the basic operation of the proposed mechanismis the same as that in the EDCA.

The proposed mechanism blocks the transmissions oflower priority ACs when higher priority one has packetsto transmit. In other words, lower priority ones do nottransmit their packets until no higher priority ones con-tend for the channel.

In the EDCA, a flow of a given AC first senses the wire-less channel medium. After sensing the idle duration ofAIFS period, the flow waits for random backoff time beforetransmitting. However, in the proposed mechanism, a flowsends a busy tone shorter than aSlotTime after sensing anidle channel for the (AIFS – aSlotTime) period. After sendingits own busy tone, the flow operates its packet transmis-sion procedure like the EDCA. So, it decreases its backoffcounter as long as the channel is sensed idle, does not de-crease when a transmission is detected on the channel, andtries to transmit a packet when the backoff counter reacheszero. On the contrary, if a flow receives a busy tone at anytime within its (AIFS – aSlotTime) period, then the flow sus-pends its busy tone transmission and backoff procedure. Inother words, the flow stops its current channel contention,and waits until a packet transmission occurs. As long as atleast one flow of higher AC exists, all flows of lower ACswill sense a busy tone within their (AIFS – aSlotTime)periods.

For the operation of the proposed mechanism, let us as-sume that, unlike the EDCA, higher priority AC always has asmaller AIFSN than lower priority one, that is AIFSN½i� >AIFSN½j� for 0 6 i < j 6 3.

The busy tone transmission time varies with the packetarrival time and channel status. In order to determine thetime, the proposed mechanism uses three parameters:the packet arrival time (PAT) at the MAC layer, the lastchannel busy time (LCBT) due to the recent packet trans-mission, and the AIFS of the lowest priority AC (LAIFS).LCBT is set to the completion time of a packet transmission.When a packet transmission collision is detected, it is setto the end time of (EIFS � DIFS). The reason is that theEDCA backoff operation starts after the channel is deter-mined to be idle for the duration of the (EIFS � DIFS + AIFS)period [4], where the EIFS is an extended inter-framespace.

There are two possible cases for determining the trans-mission time of a busy tone (see Fig. 1). If a flow receives abusy tone between LCBT and PAT, it defers its operationuntil it senses the channel busy by a packet transmission.Otherwise, it operates as in the following two cases. First,in case that a flow receives a new packet from the upperlayer before LCBT, it sends its busy tone after sensing anidle channel for (AIFS – aSlotTime) period (see Fig. 1a). Sec-ond, when a packet arrives after LCBT, the channel shouldbe determined to be idle until (LCBT + LAIFS � N +AIFS � aSlotTime) before the busy tone is allowed to trans-

Busy Medium

AIFS

Time

Packet arrival

LCBT

Slot TimeBusy Tone

Backoff Slots

Busy Medium

LAIFS

Time

AIFS

Packet arrival

LCBT

BackoffSlots

Fig. 1. Busy tone transmission time according to the packet arrival time.

New Packet Arrival

NAV = 0 ?

PAT > LCBT ?

Listen forAIFS - aSlotTime

Defer untilLCBT+LAIFS* N

Save LBT,Defer until channel busyRx Busy Tone ?

Tx Busy Tone

Decrease Backoff

Channel idle ?

Tx DATA

N

Y

Y

N

Y

N

Y

N Save LCBT

Success ?

Save LCBT,Reset CW to CWmin

Increase CW

Y

N

LBT > LCBT ? Y

N

PAT: Packet Arrival Time

LCBT: Last Channel Busy TimeCW: Contention Window

LAIFS: AIFS of Lowest priority traffic

LBT: Last Busy Tone Rx timeNAV: Network Allocation Vector

N : Alignment Integer

Fig. 2. Operation procedure of the priority access mechanism.

6 S. Kim et al. / Computer Networks 54 (2010) 3–12

mit (see Fig. 1b), where N is used to align the start time ofAIFS period with the integral multiples of LAIFS, and is

PAT�LCBTLAIFS

� �. dxe rounds to the smallest integer greater than

or equal to x. Even though a flow has no packets to trans-mit, the proposed scheme requires the flow to measurethe time passed after the busy medium in the unit of LAIFSso that it can know when to start its own AIFS sense time.This alignment is needed to ensure the channel contentionto be among flows with the same priority AC. If a flowsends its busy tone after sensing the idle duration of (AIFS– aSlotTime) period without the alignment, it may cause abusy tone collision with flows of other priority ACs. Then,the flows with the collided busy tone contend for the chan-nel at the same time so that there exist effects among dif-ferent priority ACs. Before the busy tone transmission timeexpires, if a busy tone is detected from other flows, then aflow should defer its operation until a packet transmissionoccurs, and operate as the first case. Fig. 2 shows the oper-ation procedure of the proposed priority accessmechanism.

Table 1System parameters.

Parameter Value

Data bit rate 54 MbpsControl bit rate 6 MbpsSlot Time 9 lsSIFS 16 lsRetry limit 7Propagation delay 1 lsMAC header 26 OctetsFCS 4 OctetsPHY PLCP preamble length 16 lsPHY PLCP header length 5 OctetsACK 14 OctetsBeacon interval 200 ms

In the priority access mechanism, distinguishing a busytone from a packet transmission is very important to guar-antee the proper operation. To do this, the duration of atransmission is used. Transmission time for a packet hasat least 3-time slot duration because it includes physicalpreamble and header of 20 ls, which is from Table 1 inSection 5. A busy tone duration is smaller than one timeslot. Estimating the duration is simple without any addi-tional overheads or costs because every station performsthe carrier sensing. Each station, by using the carrier sens-ing, observes the channel status and measures the durationof busy period. Therefore, the proposed mechanism candistinguish between busy tone and packet transmissionwhen receiving a signal.

In [18,19], the authors proposed a deterministic prior-ity access scheme, which is similar to our proposed mech-anism. However, their scheme is different from ours inthe following way. It provides the deterministically prior-itized access only to voice traffic over data traffic. A flowsends a busy tone instead of its backoff procedure aftersensing an idle channel for AIFS period. The length ofthe busy tone is equal to its backoff counter. When theflow completes the transmission of the busy tone, itchecks the channel status. If the channel is busy, theflow defers the current contention. Otherwise, the flow

S. Kim et al. / Computer Networks 54 (2010) 3–12 7

transmits its packets. In this scheme, a packet is transmit-ted by a flow with the longest busy tone. If there is atleast one voice flow, all data flows will sense the busytone within their AIFS period, and defer their channelcontentions.

4. Admission control

In this section, we propose an admission control mech-anism, which measures some metrics characterizing thewireless channel based on the priority access and thenmakes an admission decision. The proposed scheme iscalled priority access-based admission control (PAAC).We consider an IEEE 802.11e BSS (Basic Service Set) withinfrastructure. The admission control module is locatedat the AP of a BSS and is based on the procedure describedin the IEEE 802.11e standard [4]. Before a new flow startsits packet transmission, it sends the AP an ADDTS (AddTraffic Stream) Request frame containing its traffic specifi-cation in order to request admission. Based on the channelstatus information measured by the AP and traffic specifi-cation information provided by the flow, the admissioncontrol module decides whether the flow can be admittedto the network without affecting the QoS of existing flows.If there is no effect, the AP sends an ADDTS Response frameindicating that the request is accepted. Otherwise, a rejec-tion response is sent back to the flow.

4.1. Channel time prediction for a new flow

In order to make a right admission decision, the APneeds to accurately predict the channel time usage aftera new flow is admitted. An admitted new flow can causenot only extra transmission time, but also more collisiontime and backoff time. In this subsection, we describehow to obtain the channel time required by a new flow.

The AP measures three kinds of channel status informa-tion for each AC i ði ¼ 0; . . . ;3Þ: average channel time usageratio Ui, average collision probability Pi, and average back-off time Bi per transmission attempt. The channel timeusage ratio is the fraction of time during which the channelis used to transmit each priority traffic. The following tworeasons explain why estimating the required informationis so simple. First, in the proposed mechanism, each ACseparately performs its own packet transmission operationso that unlike the previous measurement-based admissioncontrol mechanisms, the proposed mechanism can recog-nize which AC flows cause collisions. Second, the AP per-forms the carrier sensing mechanism without anyadditional overheads or costs.

Note that we use the collision probability to compen-sate transmission failures. We can exactly measure theprobability for each traffic priority based on the priority ac-cess. In [13,14], the authors allocated the channel timebased on successful transmission time and SurplusFactor,since they do not recognize which AC flows cause colli-sions. Therefore, their scheme may make wrong admissiondecisions.

We assume the channel status information is updatedat the end of each beacon interval. For each AC i, the AP ob-

serves the channel status and measures the duration oftransmission interval TIi, total backoff time BPi, and thenumber of transmissions NTi and collided transmissionsNCi during a beacon interval BI. Transmission interval isthe time duration between two consecutive packet trans-missions, which is made up of three components: AIFS,backoff, and packet transmission. At the end of a beaconinterval, instantaneously measured channel time usage ra-tio Umeasure

i , collision probability Pmeasurei , and average back-

off time upon a transmission Bmeasurei are calculated as

follows.

Umeasurei ¼ TIi

BI; ð3Þ

Pmeasurei ¼ NCi

NTi; ð4Þ

Bmeasurei ¼ BPi

NTi: ð5Þ

From (3)–(5), Ui; Pi, and Bi are approximated by exploitingthe following moving averaging window:

Ui ¼ aUi þ ð1� aÞUmeasurei ; ð6Þ

Pi ¼ aPi þ ð1� aÞPmeasurei ; ð7Þ

Bi ¼ aBi þ ð1� aÞBmeasurei ; ð8Þ

where a is a smoothing factor in the range of [0,1]. In [20],the impact of a on the performance has been studied. The avalue close to 1 makes the reflection of the changed net-work conditions significantly longer. When the value issmall, the fluctuations of the network conditions affectthe performance remarkably. Thus in [20], the authors con-cluded that the choice of 0.9 as smoothing factor is appro-priate when the network condition changes. Therefore,throughout this paper, we use the smoothing factor of 0.9.

In order to make a request, a new flow k of AC i trans-mits an ADDTS Request frame with the following parame-ters: mean packet size Li;k, mean data rate qi;k, and physicaltransmission rate Ri;k. On receiving an ADDTS Requestframe, the AP computes the channel time usage ratio re-quired by the flow. Let Ni;k be the number of packets arriv-ing at a new flow k during one beacon interval,

Ni;k ¼qi;kBI

Li;k: ð9Þ

When there are collisions during transmissions, the flowretransmits its collided packets. Therefore, required chan-nel time should include the packets to be retransmitted.Total number of packets to be transmitted is given by

Ni;k;total ¼Ni;k

1� Pi: ð10Þ

The channel time required to transmit Ni;k;total packets is asfollows:

ChTimei;k ¼ Ni;kTs þ ðNi;k;total � Ni;kÞTc þ Ni;k;totalðBi þ AIFS½i�Þ;ð11Þ

where Tc and Ts are the average time intervals duringwhich the channel is sensed busy due to collision and suc-cessful transmission, respectively. For the basic and RTS/CTS access methods, Tc and Ts are given by

8 S. Kim et al. / Computer Networks 54 (2010) 3–12

Tc;basic¼HþLþEIFS�DIFS;

Ts;basic¼HþLþSIFSþACKþ2d;

�ð12Þ

Tc;rts=cts¼RTSþEIFS�DIFS;

Ts;rts=cts¼RTSþSIFSþCTSþSIFSþHþLþSIFSþACKþ4d;

�

ð13Þ

where Hð¼ PHYhdr þMAChdrÞ is the time to transmit apacket header, d is propagation delay, SIFS is the SIFS timespace, RTS,CTS, and ACK are the times to transmit RTS, CTS,and ACK frames, respectively. L is the packet transmissiontime, which is the ratio of Li;k and Ri;k.

The channel time usage ratio required by the new flowis given by

Ui;k;require ¼ChTimei;k

BI: ð14Þ

0.4

0.6

0.8

1.0

Data

Video

l Tim

e U

sage

Rat

io

Estimated Ratio Allocated Ratio

Table 2Traffic parameters.

Parameter Voice Video Data

AIFSN 2 4 7CWmin 7 15 31CWmax 15 31 1023Packet size (Octets) 120 1000 1500Inter arrival time (ms) 10 12.5 12.5Sending rate (Kbps) 96 640 960Channel time (%) 25 40 35

4.2. Admission decision

In the priority access mechanism, each priority trafficseparately performs its own packet transmission opera-tion. In order to make admission decisions according tothe traffic type, the channel time is allocated to each traffictype (see Fig. 3). Let C be the total channel time in whichf1C portion is reserved for voice traffic, f3C portion is for vi-deo traffic, f2C portion is for sharing between voice and vi-deo traffics, and f4C portion is for data traffic ðf1 þ f2þf3 þ f4 ¼ 1Þ. These portions could be changed based onthe traffic status among voice, video, and data. Previousschemes cannot allocate the channel time to each trafficexactly due to collisions among different AC traffics. How-ever, the proposed scheme can do it since collisions amongdifferent AC traffics can be eliminated by the priority ac-cess mechanism. Partitioning the channel time ensuresthat voice or/and video traffics do not monopolize thechannel time, and data traffic can avoid performance star-vation. Data traffic uses the channel time unused by voiceor/and video traffics for high channel utilization.

On receiving a new ADDTS Request frame, the AP makesa decision whether or not to accept the flow. For a newflow k of AC i, we have the following decision criterion:

Ui þ Ui;k;require 6 CTi; ð15Þ

where CTi is the channel time usage ratio which can beused for AC i at the moment when the new request is re-ceived. This value varies dynamically with time since voiceand video traffics share f2 of the channel time, and datatraffic uses the channel time unused by voice or/and videotraffics. CTi is ðf1 þ f2 þ f3 �maxðUvideo; f3ÞÞ for voice traffic,ðf1 þ f2 þ f3 �maxðUvoice; f1ÞÞ for video traffic, andð1� Uvoice � UvideoÞ for data traffic.

If (15) is satisfied, the new flow request is accepted.Otherwise, it is denied.

Voice Voice/Video Video Data

Total Channel Time (C)

f1 C f 2 C f 3 C f 4 C

Fig. 3. Channel time allocation for each traffic type.

5. Performance evaluation

In this section, we discuss the performance of the pro-posed scheme. We have implemented the proposedscheme with the NS-2 simulator. System parameters usedin the simulation are listed in Table 1. We simulated anIEEE 802.11a network with transmission rates of 54 Mbpsfor data packets and of 6 Mbps for control packets, respec-tively. We have three types of traffics: voice, video, anddata. Traffic parameters are listed in Table 2. A constantbit rate (CBR) model is used for all three traffics. Weallocate 65% of the channel time to real-time traffics(i.e., f1 ¼ ATL½3� ¼ 0:25; f2 ¼ 0:0; f3 ¼ ATL½2� ¼ 0:4, and f4 ¼ATL½1� ¼ 0:35). f2 is not allocated for performance compar-ison with the DAC scheme. Data traffic is allowed to use thechannel time unused by real-time traffics.

In the proposed scheme, the number of flows contend-ing with each other at the same time is fewer than thestandard MAC since each priority traffic separately con-tends for the channel, so that CWmin and CWmax can beset to smaller values. For the proposed scheme, we set(CWmin,CWmax) to (3,7), (7,15), and (15,511) for voice, vi-deo, and data traffics, respectively.

For the DAC scheme, we have SurplusFactor ¼ 1:2;/ ¼ 1;m ¼ 1;u ¼ 1; h ¼ 1:5, and riþ1 ¼ 2riði ¼ 1; . . . ;

Lretry � 1Þ [13].In the simulation, we consider a single hop wireless LAN

with basic access method and only uplink traffics. We as-sume that each station has a single flow of voice, video, ordata traffic, respectively. The simulation time is 120 s.

0 5 10 15 20 25 30 0.0

0.2

Voice

Cha

nne

Number of Flows per Traffic

Fig. 4. Channel time usage ratio required for a new flow.

S. Kim et al. / Computer Networks 54 (2010) 3–12 9

Main performance metrics of interest are throughput,delay, and channel time usage ratio. Delay is the timeelapsed from the moment a packet arrives at the MAC layerqueue until the packet is successfully transmitted to the in-tended station including the receipt of acknowledgement.

Fig. 4 shows the channel time usage ratio obtained from(14) for a new flow. We do not use the admission controlmechanism so that the network accepts every new flow.New flows for voice, video, and data traffics periodically ar-rive to the network every 10 s at the same time. In this fig-ure, Estimated Ratio means the summation of the ratio

0 20 40 60 80 100 120 0.0

0.2

0.4

0.6

0.8

1.0

Cha

nnel

Tim

e U

sage

Rat

io

Time (s)

Voice Data Video Total

Fig. 5. Channel time usage ratio according to the simulation time.

0 20 40 600.00

0.01

0.02

0.03

0.04

0 20 40 600.0

0.1

0.2

0.3

0.4

0.5

Voice

TxLi

mit

& Tx

Use

d

Time

TxUsed

TxUsed

TxLimit

Video

Voice

Video

TxLi

mit

& Tx

Use

d

Time

Fig. 6. TxLimit and TxUsed accord

measured when k� 1 flows contend for the channel andthe ratio obtained from (14) for a new kth flow. AllocatedRatio is the usage ratio measured after the new kth flowis admitted. From this figure, we know that the channeltime usage ratio almost linearly increases and has a goodcoincidence under the stable status. At high loads, the ratioresults do not match because higher priority traffics takethe channel time from lower priority traffics so that thechannel time for lower traffics decreases. This is ignorablesince a new flow will be blocked by the proposed admis-sion control mechanism when there is no available channeltime. This figure demonstrates that the channel time usageratio obtained from (14) for a new flow is reasonable, andthe proposed scheme is able to make right admissiondecisions.

Figs. 5–8 show the performance results when theadmission control mechanism is used. The arrival patternof each traffic is as follows. At the beginning, there are noflows in the network. A new flow periodically arrives tothe network every 6 s in the order of 2 s, 4 s, or 6 s forvoice, video, or data traffics, respectively. In other words,a voice flow arrives at 2 s, 8 s, 14 s, and so on, a video flowat 4 s, 10 s, 16 s, and so on, and a data flow at 6 s, 12 s, 18 s,and so on. Furthermore, the start time of each flow is de-layed by a random time chosen from an uniform distribu-tion in the range of [0 s,0.5 s].

Fig. 5 depicts the channel time usage ratio for theproposed scheme. At light loads, data traffic with higher

80 100 120

80 100 120

(s)

TxLimit

(s)

ing to the simulation time.

0 20 40 60 80 100 120 0.0

0.5

1.0

1.5

2.0

Voice

Video

Del

ay (m

s)

Time (s)

DAC PAAC

Fig. 8. Delay according to the simulation time.

0 20 40 60 80 100 120 0

2

4

6

8

10

12

14

Voice

Video

Thro

ughp

ut (M

bps)

Time (s)

DAC PAAC

0 20 40 60 80 100 120 0

2

4

6

8

10

12

14

Thro

ughp

ut (M

bps)

Time (s)

DAC PAAC

Fig. 7. Throughput according to the simulation time.

10 S. Kim et al. / Computer Networks 54 (2010) 3–12

sending rate uses larger channel time than the other traf-fics. However, as the number of flows increases, the por-tion of channel time used by data traffic decreases, sincein the proposed scheme higher priority traffics can gainexclusive channel access over lower priority ones. In other

words, to guarantee the QoS requirement for higher prior-ity traffics, the proposed scheme discriminates lower traf-fics, and reallocates the amount of saved channel time tohigher priority traffics. From this figure, we can observethat real-time traffics use as much as the channel timeallocated in Table 2. This indicates that the starvation ofdata traffic can be avoided.

Fig. 6 shows TxLimit and TxUsed of the first admittedflows for voice and video traffics of the DAC scheme. In thisfigure, the unit of Y-axis is the channel time ratio instead ofthe amount of channel time. For a better readability, weshow two figures with a different scale of Y-axis. OnceTXOPBudget is depleted, TxLimit converges to TxSuc-cess � SurplusFactor. This ensures that a station can con-tinue consuming the same amount of channel time forsubsequent beacon intervals. TxUsed is limited by TxLimitsince a station is prevented from sending a packet whenTxUsed exceeds TxLimit. This can lead to the performancedegradation. As the number of flows becomes larger, colli-sion probability increases so that traffics need more chan-nel time to provide QoS. However, the network cannotprovide channel time enough to real-time traffics due toTxLimit based on a constant SurplusFactor. To provide morechannel time, SurplusFactor should be larger, thus resultingin fewer admitted flows. From this figure, we can observethat it is not easy to decide a globally optimal SurplusFactorvalue.

Fig. 7 shows the throughput behavior. For the proposedscheme, it shows almost the same patterns as Fig. 5. Wesee that there are no performance differences betweenthe DAC and PAAC schemes at light loads. For the PAACscheme, the throughput for voice and video traffics is sta-ble and always met with their requirements regardless ofnetwork loads. However, data traffic fluctuates at highloads since voice and video traffics use larger channel timefor retransmissions, when they have collided packets. Forthe DAC scheme, the throughput of video traffic does notremain stable. At high loads, video traffic needs more chan-nel time due to collisions, but it cannot use larger channeltime due to TxLimit as shown in Fig. 6. Therefore, the DACscheme does not meet the throughput requirements, espe-cially from the point of intersection between TxLimit andTxUsed in Fig. 6.

Fig. 8 demonstrates the delay performance. We can seethat the delay is kept low at light loads. As loads becomehigher, the proposed scheme outperforms the DAC for allpriority traffics. For the PAAC scheme, note that the in-crease of delay is not due to lower priority traffics. Themain reason of voice traffic is that the number of voiceflows contending for the channel increases. However, forvideo traffic, it is because the performance is affected bythe number of voice flows as well as its own number offlows. For the DAC scheme, the delay increases sharply.This is from the fact that in the DAC all flows always tryto access the channel and make collisions with oneanother.

As illustrated in Figs. 5–8, the proposed scheme totallyaccepts 15 voice flows, 15 video flows, and 12 data flows.The DAC scheme accepts 15 voice flows and 14 video flows.Although both schemes have the same traffic pattern andmaximum channel time for each traffic, the PAAC scheme

0 20 40 60 80 100 120 0

1

2

3

0 20 40 60 80 100 120 0

1

2

3

Del

ay (m

s)

Time (s)

PAAC

Del

ay (m

s)

Time (s)

DAC

Fig. 9. Delay of a voice flow according to traffic loads.

S. Kim et al. / Computer Networks 54 (2010) 3–12 11

accepts 1 video flow more than the DAC scheme. This indi-cates that the PAAC scheme is efficient to accommodatemore traffic. For the PAAC scheme, after 92 s, no moreflows are admitted. When the 13th new data flow arrivesat 78 s, about 98% of the channel time is occupied by theadmitted flows (i.e., about 20%, 34%, and 44% by voice, vi-deo, and data, respectively). Therefore, the new data flowis denied. Similarly, at 92 s and 94 s, voice and video traf-fics use about 24% of the channel time and about 39%respectively, so that the network cannot admit the 16thnew voice and video flows. For the DAC scheme, at 88 sand 92 s, TXOPBudeget is less than the budget required bya new flow for video and voice traffics, respectively. There-fore, the new flows are rejected.

Fig. 9 shows the delay performance of a voice flowaccording to traffic loads. To validate that the proposedscheme can solve the priority reversal problem, we simu-lated under the environment where the arrival pattern isthe same as the one mentioned above except that thereis only one voice flow. We increase the channel time for vi-deo traffic to accept up to 20 video flows. For the DAC, thedelay becomes worse as traffic loads become higher sinceall priority flows always try to access the channel andmake collisions with one another. For the PAAC, the delayremains almost the same and is always met with voicetraffic’s requirement under wide range of loads becausethe proposed scheme ensures that flows of the highestAC always access the channel first. However, at mediumloads, it slightly increases compared to light loads. This isfrom the fact that, in the proposed scheme, voice flowswith a new packet arrival during transmission intervalsof video or data flows must wait until sensing a packettransmission.

6. Conclusion

IEEE 802.11e EDCA does not ensure the QoS require-ments of multimedia applications, and causes the perfor-mance starvation of lower priority traffics at high loads.In this paper, we proposed a measurement-based admis-sion control scheme to improve the QoS performance.We first propose a priority access mechanism which usesa busy tone. Each priority traffic separately contends forthe channel. Therefore, the AP can exactly measure thechannel status for each priority traffic. And then the mea-surement information is used in the admission controlmechanism. Simulation results show that the proposedscheme is very effective and ensures the QoS requirementsof multimedia applications such as voice and video traffics,and protects the performance starvation of data traffic.

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Sunmyeng Kim received the B.S., M.S., andPh.D. degrees in information and communi-cation from Ajou University, Suwon, Korea, in2000, 2002, and 2006, respectively. From May2006 to February 2008, he was a PostdoctoralResearcher in electrical and computer engi-neering with the University of Florida,Gainesville. In March 2008, he then joined theSchool of Computer and Software Engineer-ing, Kumoh National Institute of Technology,Gumi, Korea, as a Full-Time Lecturer. Hisresearch interests include resource manage-

ment, wireless LANs and PANs, wireless mesh networks, and quality-of-service enhancement.

Young-Jong Cho received the B.S. degree inElectronic Engineering from Seoul National

University, in 1983 and M.S. and Ph.D. degreesin Electronic Engineering from KAIST in 1985and 1989, respectively. Since 1996, he hasbeen a professor of the Department of Infor-mation and Computer at Ajou University.From 1985 to 1996, he had been a principalengineer at LG Co., Korea. In 1993, he joinedthe AT&T Bell Lab. as a researcher and in 2003,he was a visiting professor at GMU. Hisresearch interests include performance anal-

ysis of broadband communication systems and development of highspeed Internet devices.

orks 54 (2010) 3–12

Yong K. Kim received the B.S. degree inElectronics Engineering from Ajou University,Korea in 1988 and the M.S.E. degree in Elec-trical and Computer Engineering from Uni-versity of Alabama in Huntsville 1993. Hereceived his Ph.D. degree in Electrical andComputer Engineering in North Carolina StateUniversity 2001. He is currently in AssociateProfessor at School of Electrical Electronicsand Information Engineering, WonkwangUniversity, Korea. His research interests arefiber optics & sensing, fiber property mea-

surement using fiber gratings, photorefractive fiber and crystal devices.