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Admission control scheme based on priority access for wireless LANs
Computer Networks 54 (2010) 3–12
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Admission control scheme based on priority access for wireless LANs q Sunmyeng Kim a,*, Young-Jong Cho b, Yong K. Kim c a b c
School of Computer and Software Engineering, Kumoh National Institute of Technology, Gumi, Republic of Korea Department of Information and Computer Engineering, Ajou University, Suwon, Republic of Korea 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 2008 Received in revised form 7 April 2009 Accepted 9 July 2009 Available online 16 July 2009 Responsible Editor: E. Gregori Keywords: EDCA IEEE 802.11e QoS Admission control Priority access
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 IEEE 802.11e standard. However, it cannot guarantee the stringent real-time constraints of multimedia applications such as voice and video without an efficient method of controlling network 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 measure the channel status per traffic type, we propose a priority access mechanism in which each priority traffic is distinguished by a busy tone, and separately performs its own packet transmission operation. Then, admission control mechanism protects existing flows from new ones, and maintains the QoS of the admitted flows based on the measured channel status information. Performance of the proposed scheme is evaluated by simulation. Our results show that the proposed scheme is very effective in guaranteeing the QoS of multimedia applications as well as in avoiding the performance starvation of low priority traffics. Ó 2009 Elsevier B.V. All rights reserved.
1. Introduction The IEEE 802.11 wireless LAN is widely used for wireless access due to its easy deployment and low cost. The IEEE 802.11 standard defines a medium access control (MAC) protocol for sharing the channel among stations [1]. The distributed coordination function (DCF) was designed for a contention-based channel access. The DCF has two data transmission methods: the default basic access and optional RTS/CTS (request-to-send/clear-to-send) access. The widespread use of multimedia applications requires new features such as high bandwidth and small average delay in wireless LANs. Unfortunately, the IEEE 802.11
q This paper was supported by Research Fund, Kumoh National Institute of Technology. * Corresponding author. Tel.: +82 54 478 7547; fax: +82 54 478 7539. E-mail addresses: [email protected] (S. Kim), [email protected] (Y.-J. Cho), [email protected] (Y.K. Kim).
1389-1286/$ - see front matter Ó 2009 Elsevier B.V. All rights reserved. doi:10.1016/j.comnet.2009.07.002
MAC protocol cannot support quality of service (QoS) requirements [2,3]. In order to support multimedia applications with tight QoS requirements in the IEEE 802.11 MAC protocol, the IEEE 802.11e has been standardized [4]. It introduces a contention-based new channel access mechanism called enhanced distributed channel access (EDCA). The EDCA supports the QoS by introducing four access categories (ACs). To differentiate the ACs, the EDCA uses a set of AC specific parameters, which include minimum contention window, maximum contention window, and arbitration inter-frame space (AIFS). In the previous work [2,5–7], the authors focused on studying the service differentiation. However, service differentiation alone is not sufficient to provide the QoS of multimedia applications without an efficient method of controlling network loads. In this paper, we focus on admission control mechanism for the IEEE 802.11e EDCA. Many admission control mechanisms for 802.11e EDCA have been proposed. The previous mechanisms are classified into two approaches: model-based and
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S. Kim et al. / Computer Networks 54 (2010) 3–12
measurement-based admission control [8]. Model-based admission control approach uses an analytical model to decide whether a new flow is admitted or rejected [9–12]. Measurement-based approach uses measurement information (e.g., throughput, collision probability, or delay) of existing traffic 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 few unpractical hypotheses to calculate the QoS metrics of all existing flows in addition to a new one. They do not reflect the characteristics of real traffics. Therefore, the modelbased mechanisms are always inaccurate and clearly not applicable to realistic environments. Second, it is hard for measurement-based schemes to make a right admission decision since the schemes cannot accurately predict the channel time usage after a new flow is admitted to the network. For example, the scheme proposed in [13,14] allocates the bandwidth to flows and makes an admission decision based on successful transmission time and a constant parameter SurplusFactor. The parameter is used to compensate potential transmission failures because this scheme does not recognize which AC flows send packets when there are collisions. The inaccuracy of this parameter can lead to making a wrong decision. Third, the previous schemes do not solve the priority reversal problem that low priority flows have a shorter backoff counter than higher priority ones, and consequently access the channel [16]. Since each flow decreases its backoff counter when the channel is idle, it will eventually have a small value although its initial counter is large. A high priority flow cannot be always assured to have a smaller backoff counter since the backoff counter is randomly selected based on the uniform distribution. Therefore, lower priority flows can transmit their packets prior to higher priority ones. As a result, higher priority traffic may wait a long time for the channel contention. Fourth, although an admission control scheme works well, higher priority traffics are vulnerable to lower priority traffics since the latter ones may cause many collisions with the former ones. Finally, most of the previous schemes first regulate data traffic in order to provide the QoS to the real-time traffics. Therefore, the performance starvation of data traffic can be caused at high 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, the priority access mechanism uses a short duration busy tone signal (i.e., pulses of energy) [17]. Each flow sends a busy tone in the last slot of its own AIFS period, and then it operates backoff procedure like the EDCA. However, flows that receive the busy tone suspend their busy tone transmission and backoff procedure. This ensures that each priority traffic separately performs its own packet transmission operation. Second, the admission control mechanism regulates network loads and protects admitted flows from new ones. The priority access mechanism is used to facilitate the admission control. The access point (AP) measures the channel status information for each traffic type, and then separately makes an admission decision. The proposed scheme makes the following contributions: (1) The priority access mechanism blocks the trans-
missions of lower priority ACs when higher priority one has packets to transmit. This ensures that flows of the highest AC always access the channel first. Therefore, the proposed scheme can solve the priority reversal problem. (2) The priority access mechanism also ensures the channel contention to be among flows with the same priority AC. In other words, there are no channel contentions and collisions among flows with different priority ACs. Therefore, the AP exactly measures the channel status information for each priority AC, and then accurately predicts the channel time usage after a new flow is admitted. (3) Bandwidth allocation to each priority AC in the previous schemes is extremely challenging due to collisions among different priority ACs. However, the proposed scheme can allocate the bandwidth to each priority AC exactly since there are no collisions among different priority ACs. Accordingly, the proposed scheme can prevent the starvation of data traffic. The paper is organized as follows. In Section 2, we give a brief introduction of the IEEE 802.11e EDCA and the scheme proposed in [13,14]. In Section 3, the proposed priority access mechanism is presented in detail. In Section 4, we describe the admission control mechanism based on the priority access. In Section 5, performance studies are carried out through simulation results. Finally, we draw a conclusion 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 introduces a new medium access method called hybrid coordination function (HCF), which combines a contention-based enhanced distributed access mechanism (EDCA) and a controlled channel access mechanism (HCCA). The EDCA is an enhanced variant of the DCF. In the DCF, all stations contend for the channel with the same priority. On the other hand, the EDCA supports several priority levels by introducing an access category (AC) concept. A station have up to four ACs to support eight user priorities. Each AC is implemented as a separate queue. Each packet arrives at the MAC layer with a priority from higher layer, and is mapped to one AC according to the priority. AC 3, AC 2, AC 1, and AC 0 are for voice, video, best effort data, and background traffics, respectively. In order to differentiate the traffics, the EDCA uses a set of AC specific parameters, which include minimum contention window (CWmin[i]), maximum contention window (CWmax[i]), and arbitration inter-frame space (AIFS[i]) for AC i ði ¼ 0; . . . ; 3Þ. The AIFS is at least distributed inter-frame space (DIFS) long and is calculated with the arbitration inter-frame space number (AIFSN[i]). The duration of AIFS[i] is defined by AIFS½i ¼ SIFS þ AIFSN½i aSlotTime, where SIFS is a short inter-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 that in the above inequalities, at least one must be ‘‘not equal to”. The EDCA assigns a smaller CWmin,CWmax, or AIFS to
S. Kim et al. / Computer Networks 54 (2010) 3–12
higher priority AC in order to provide the higher chance of accessing the channel. Therefore, in the EDCA, the support of QoS can be achieved by differentiating the probability of accessing 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]. At the first level, the admitted voice and video flows are protected from new and other voice and video flows. The AP measures the amount of channel time (TxTime[i]) occupied by successful transmissions from each AC i during a beacon period. And then, it calculates the additional amount of channel time (TXOPBudget[i]) available for AC i during the next beacon interval.
TXOPBudget½i ¼ maxðATL½i TxTime½i SurplusFactor½i; 0Þ;
ð1Þ
where ATL[i] is the maximum amount of channel time for AC i per beacon interval, and SurplusFactor[i] is used to compensate potential transmission failures due to collisions. The budget information is delivered to each station via a beacon frame. Each station maintains the following variables for AC i: TxUsed[i], TxSuccess[i], TxLimit[i], TxRemainder[i], and TxMemory[i]. TxUsed[i] is the amount of channel time occupied by transmissions, regardless of success or not, from AC i of this station. TxSuccess[i] is the amount of channel time occupied by successful transmissions. A station does not transmit a data packet when TxUsed[i] exceeds TxLimit[i], where how to determine this value is described below. If a station is prevented from sending a frame for this reason, TxRemainder½i ¼ TxLimit½i TxUsed½i. Otherwise, TxRemainder½i ¼ 0. TxMemory[i] memorizes the amount of channel time that AC i of this station utilized during a beacon interval, and is TxMemory½i ¼ a TxMemory½i þ ð1 aÞ ðTxSuccess½i SurplusFactor½iþ TXOPBudget½iÞ. a is a smoothing factor. TxLimit[i] is defined by 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 of AC i, and / is a fraction. If this equation is satisfied, the new flow is admitted. At the second level, the voice and video flows are protected from best-effort data traffic. In order to control data transmissions according to traffic conditions, the DAC scheme dynamically adjusts the EDCA parameters for data traffic. This scheme increases the contention window size much faster than the standard by using a larger contention window increasing factor ri for the backoff stage i instead of 2 used in the standard when a collision occurs. That is ri < rj ðr1 P 2Þ for 1 6 i < j 6 Lretry , where Lretry is the retry limit. Whenever a data frame reaches the retry limit, CWmin and AIFS increase (i.e., CWmin ¼ h CWminðh > 1Þ and AIFS ¼ w AIFSðw > 1Þ). Whenever a station successfully transmits m consecutive packets, CWmin and AIFS decrease (i.e., CWmin ¼ CWmin=hðh > 1Þ and AIFS ¼ AIFS=wðw > 1Þ).
5
3. Priority access In this section we describe the priority access mechanism that ensures the channel contention-based on traffic priority. We consider a single hop IEEE 802.11e network. Our mechanism needs minor modifications to the EDCA. However, the basic operation of the proposed mechanism is the same as that in the EDCA. The proposed mechanism blocks the transmissions of lower priority ACs when higher priority one has packets to transmit. In other words, lower priority ones do not transmit their packets until no higher priority ones contend for the channel. In the EDCA, a flow of a given AC first senses the wireless channel medium. After sensing the idle duration of AIFS period, the flow waits for random backoff time before transmitting. However, in the proposed mechanism, a flow sends a busy tone shorter than aSlotTime after sensing an idle channel for the (AIFS – aSlotTime) period. After sending its own busy tone, the flow operates its packet transmission procedure like the EDCA. So, it decreases its backoff counter as long as the channel is sensed idle, does not decrease when a transmission is detected on the channel, and tries to transmit a packet when the backoff counter reaches zero. On the contrary, if a flow receives a busy tone at any time within its (AIFS – aSlotTime) period, then the flow suspends its busy tone transmission and backoff procedure. In other words, the flow stops its current channel contention, and waits until a packet transmission occurs. As long as at least one flow of higher AC exists, all flows of lower ACs will sense a busy tone within their (AIFS – aSlotTime) periods. For the operation of the proposed mechanism, let us assume that, unlike the EDCA, higher priority AC always has a smaller AIFSN than lower priority one, that is AIFSN½i > AIFSN½j for 0 6 i < j 6 3. The busy tone transmission time varies with the packet arrival time and channel status. In order to determine the time, the proposed mechanism uses three parameters: the packet arrival time (PAT) at the MAC layer, the last channel busy time (LCBT) due to the recent packet transmission, 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 set to the end time of (EIFS DIFS). The reason is that the EDCA backoff operation starts after the channel is determined to be idle for the duration of the (EIFS DIFS + AIFS) period [4], where the EIFS is an extended inter-frame space. There are two possible cases for determining the transmission time of a busy tone (see Fig. 1). If a flow receives a busy tone between LCBT and PAT, it defers its operation until 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 upper layer before LCBT, it sends its busy tone after sensing an idle channel for (AIFS – aSlotTime) period (see Fig. 1a). Second, when a packet arrives after LCBT, the channel should be determined to be idle until (LCBT + LAIFS N + AIFS aSlotTime) before the busy tone is allowed to trans-
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S. Kim et al. / Computer Networks 54 (2010) 3–12
LCBT
New Packet Arrival
AIFS Busy Medium
Backoff Slots
NAV = 0 ?
Time
Y LBT > LCBT ?
Packet arrival
N
Y
N
Busy Tone
PAT > LCBT ?
Slot Time
Y
Defer until LCBT+LAIFS* N
Y
Save LBT, Defer until channel busy
N Listen for AIFS - aSlotTime
LCBT
Rx Busy Tone ?
LAIFS
AIFS
N
Backoff Slots
Busy Medium
Tx Busy Tone
Time Decrease Backoff
Packet arrival Channel idle ?
N
Save LCBT
Y Fig. 1. Busy tone transmission time according to the packet arrival time.
Tx DATA
mit (see Fig. 1b), where N is used to align the start time of AIFS period with the integral multiples of LAIFS, and is PATLCBT . dxe rounds to the smallest integer greater than LAIFS or equal to x. Even though a flow has no packets to transmit, the proposed scheme requires the flow to measure the time passed after the busy medium in the unit of LAIFS so that it can know when to start its own AIFS sense time. This alignment is needed to ensure the channel contention to be among flows with the same priority AC. If a flow sends its busy tone after sensing the idle duration of (AIFS – aSlotTime) period without the alignment, it may cause a busy tone collision with flows of other priority ACs. Then, the flows with the collided busy tone contend for the channel at the same time so that there exist effects among different priority ACs. Before the busy tone transmission time expires, if a busy tone is detected from other flows, then a flow should defer its operation until a packet transmission occurs, and operate as the first case. Fig. 2 shows the operation procedure of the proposed priority access mechanism.
Success ?
Table 1 System parameters. Parameter
Value
Data bit rate Control bit rate Slot Time SIFS Retry limit Propagation delay MAC header FCS PHY PLCP preamble length PHY PLCP header length ACK Beacon interval
54 Mbps 6 Mbps 9 ls 16 ls 7 1 ls 26 Octets 4 Octets 16 ls 5 Octets 14 Octets 200 ms
N
Increase CW
Y Save LCBT, Reset CW to CWmin LAIFS: AIFS of Lowest priority traffic LCBT: Last Channel Busy Time NAV: Network Allocation Vector LBT: Last Busy Tone Rx time CW: Contention Window PAT: Packet Arrival Time N : Alignment Integer Fig. 2. Operation procedure of the priority access mechanism.
In the priority access mechanism, distinguishing a busy tone from a packet transmission is very important to guarantee the proper operation. To do this, the duration of a transmission is used. Transmission time for a packet has at least 3-time slot duration because it includes physical preamble and header of 20 ls, which is from Table 1 in Section 5. A busy tone duration is smaller than one time slot. Estimating the duration is simple without any additional overheads or costs because every station performs the carrier sensing. Each station, by using the carrier sensing, observes the channel status and measures the duration of busy period. Therefore, the proposed mechanism can distinguish between busy tone and packet transmission when receiving a signal. In [18,19], the authors proposed a deterministic priority access scheme, which is similar to our proposed mechanism. However, their scheme is different from ours in the following way. It provides the deterministically prioritized access only to voice traffic over data traffic. A flow sends a busy tone instead of its backoff procedure after sensing an idle channel for AIFS period. The length of the busy tone is equal to its backoff counter. When the flow completes the transmission of the busy tone, it checks the channel status. If the channel is busy, the flow defers the current contention. Otherwise, the flow
S. Kim et al. / Computer Networks 54 (2010) 3–12
transmits its packets. In this scheme, a packet is transmitted by a flow with the longest busy tone. If there is at least one voice flow, all data flows will sense the busy tone within their AIFS period, and defer their channel contentions.
4. Admission control In this section, we propose an admission control mechanism, which measures some metrics characterizing the wireless channel based on the priority access and then makes an admission decision. The proposed scheme is called priority access-based admission control (PAAC). We consider an IEEE 802.11e BSS (Basic Service Set) with infrastructure. The admission control module is located at the AP of a BSS and is based on the procedure described in the IEEE 802.11e standard [4]. Before a new flow starts its packet transmission, it sends the AP an ADDTS (Add Traffic Stream) Request frame containing its traffic specification in order to request admission. Based on the channel status information measured by the AP and traffic specification information provided by the flow, the admission control module decides whether the flow can be admitted to the network without affecting the QoS of existing flows. If there is no effect, the AP sends an ADDTS Response frame indicating that the request is accepted. Otherwise, a rejection 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 AP needs to accurately predict the channel time usage after a new flow is admitted. An admitted new flow can cause not only extra transmission time, but also more collision time and backoff time. In this subsection, we describe how to obtain the channel time required by a new flow. The AP measures three kinds of channel status information for each AC i ði ¼ 0; . . . ; 3Þ: average channel time usage ratio U i , average collision probability P i , and average backoff time Bi per transmission attempt. The channel time usage ratio is the fraction of time during which the channel is used to transmit each priority traffic. The following two reasons explain why estimating the required information is so simple. First, in the proposed mechanism, each AC separately performs its own packet transmission operation so that unlike the previous measurement-based admission control mechanisms, the proposed mechanism can recognize which AC flows cause collisions. Second, the AP performs the carrier sensing mechanism without any additional overheads or costs. Note that we use the collision probability to compensate transmission failures. We can exactly measure the probability for each traffic priority based on the priority access. In [13,14], the authors allocated the channel time based on successful transmission time and SurplusFactor, since they do not recognize which AC flows cause collisions. Therefore, their scheme may make wrong admission decisions. We assume the channel status information is updated at the end of each beacon interval. For each AC i, the AP ob-
7
serves the channel status and measures the duration of transmission interval TIi , total backoff time BP i , and the number of transmissions NT i and collided transmissions NC i during a beacon interval BI. Transmission interval is the time duration between two consecutive packet transmissions, which is made up of three components: AIFS, backoff, and packet transmission. At the end of a beacon interval, instantaneously measured channel time usage ra, collision probability P measure , and average backtio U measure i i are calculated as off time upon a transmission Bmeasure i follows.
TIi ; BI NC i ¼ ; NT i BPi ¼ : NT i
U measure ¼ i
ð3Þ
P measure i
ð4Þ
Bmeasure i
ð5Þ
From (3)–(5), U i ; Pi , and Bi are approximated by exploiting the following moving averaging window:
U i ¼ aU i þ ð1 aÞU measure ; i
ð6Þ
; P i ¼ aPi þ ð1 aÞP measure i
ð7Þ
; Bi ¼ aBi þ ð1 aÞBmeasure i
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 a value close to 1 makes the reflection of the changed network conditions significantly longer. When the value is small, the fluctuations of the network conditions affect the performance remarkably. Thus in [20], the authors concluded that the choice of 0.9 as smoothing factor is appropriate 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 transmits an ADDTS Request frame with the following parameters: mean packet size Li;k , mean data rate qi;k , and physical transmission rate Ri;k . On receiving an ADDTS Request frame, the AP computes the channel time usage ratio required by the flow. Let N i;k be the number of packets arriving at a new flow k during one beacon interval,
Ni;k ¼
qi;k BI Li;k
:
ð9Þ
When there are collisions during transmissions, the flow retransmits its collided packets. Therefore, required channel time should include the packets to be retransmitted. Total number of packets to be transmitted is given by
Ni;k;total ¼
Ni;k : 1 Pi
ð10Þ
The channel time required to transmit N i;k;total packets is as follows:
ChTimei;k ¼ Ni;k T s þ ðNi;k;total Ni;k ÞT c þ N i;k;total ðBi þ AIFS½iÞ; ð11Þ where T c and T s are the average time intervals during which the channel is sensed busy due to collision and successful transmission, respectively. For the basic and RTS/ CTS access methods, T c and T s are given by
S. Kim et al. / Computer Networks 54 (2010) 3–12
T c;basic ¼ H þ L þ EIFS DIFS;
ð12Þ
T s;basic ¼ H þ L þ SIFS þ ACK þ 2d; T c;rts=cts ¼ RTS þ EIFS DIFS;
T s;rts=cts ¼ RTS þ SIFS þ CTS þ SIFS þ H þ L þ SIFS þ ACK þ 4d; ð13Þ where Hð¼ PHYhdr þ MAChdrÞ is the time to transmit a packet header, d is propagation delay, SIFS is the SIFS time space, RTS,CTS, and ACK are the times to transmit RTS, CTS, and ACK frames, respectively. L is the packet transmission time, which is the ratio of Li;k and Ri;k . The channel time usage ratio required by the new flow is given by
U i;k;require ¼
ChTimei;k : BI
ð14Þ
4.2. Admission decision In the priority access mechanism, each priority traffic separately performs its own packet transmission operation. In order to make admission decisions according to the traffic type, the channel time is allocated to each traffic type (see Fig. 3). Let C be the total channel time in which f1 C portion is reserved for voice traffic, f3 C portion is for video traffic, f2 C portion is for sharing between voice and video traffics, and f4 C portion is for data traffic ðf1 þ f2 þ f3 þ f4 ¼ 1Þ. These portions could be changed based on the traffic status among voice, video, and data. Previous schemes cannot allocate the channel time to each traffic exactly due to collisions among different AC traffics. However, the proposed scheme can do it since collisions among different AC traffics can be eliminated by the priority access mechanism. Partitioning the channel time ensures that voice or/and video traffics do not monopolize the channel time, and data traffic can avoid performance starvation. Data traffic uses the channel time unused by voice or/and video traffics for high channel utilization. On receiving a new ADDTS Request frame, the AP makes a decision whether or not to accept the flow. For a new flow k of AC i, we have the following decision criterion:
U i þ U i;k;require 6 CT i ;
ð15Þ
where CT i is the channel time usage ratio which can be used for AC i at the moment when the new request is received. This value varies dynamically with time since voice and video traffics share f2 of the channel time, and data traffic uses the channel time unused by voice or/and video traffics. CT i is ðf1 þ f2 þ f3 maxðU video ; f3 ÞÞ for voice traffic, ðf1 þ f2 þ f3 maxðU voice ; f1 ÞÞ for video traffic, and ð1 U voice U video Þ for data traffic. If (15) is satisfied, the new flow request is accepted. Otherwise, it is denied. Total Channel Time (C) f1 C
f2 C
f3 C
f4 C
Voice
Voice/Video
Video
Data
5. Performance evaluation In this section, we discuss the performance of the proposed scheme. We have implemented the proposed scheme with the NS-2 simulator. System parameters used in the simulation are listed in Table 1. We simulated an IEEE 802.11a network with transmission rates of 54 Mbps for data packets and of 6 Mbps for control packets, respectively. We have three types of traffics: voice, video, and data. Traffic parameters are listed in Table 2. A constant bit rate (CBR) model is used for all three traffics. We allocate 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 ¼ ATL½1 ¼ 0:35). f2 is not allocated for performance comparison with the DAC scheme. Data traffic is allowed to use the channel time unused by real-time traffics. In the proposed scheme, the number of flows contending with each other at the same time is fewer than the standard MAC since each priority traffic separately contends for the channel, so that CWmin and CWmax can be set to smaller values. For the proposed scheme, we set (CWmin, CWmax) to (3, 7), (7, 15), and (15, 511) for voice, video, and data traffics, respectively. For the DAC scheme, we have SurplusFactor ¼ 1:2; and riþ1 ¼ 2ri ði ¼ 1; . . . ; / ¼ 1; m ¼ 1; u ¼ 1; h ¼ 1:5, Lretry 1Þ [13]. In the simulation, we consider a single hop wireless LAN with basic access method and only uplink traffics. We assume that each station has a single flow of voice, video, or data traffic, respectively. The simulation time is 120 s.
Table 2 Traffic parameters. Parameter
Voice
Video
Data
AIFSN CWmin CWmax Packet size (Octets) Inter arrival time (ms) Sending rate (Kbps) Channel time (%)
2 7 15 120 10 96 25
4 15 31 1000 12.5 640 40
7 31 1023 1500 12.5 960 35
1.0
Channel Time Usage Ratio
8
Estimated Ratio Allocated Ratio
0.8
Video
0.6 Data
0.4 Voice
0.2
0.0 0
5
10
15
20
25
Number of Flows per Traffic Fig. 3. Channel time allocation for each traffic type.
Fig. 4. Channel time usage ratio required for a new flow.
30
9
S. Kim et al. / Computer Networks 54 (2010) 3–12
Main performance metrics of interest are throughput, delay, and channel time usage ratio. Delay is the time elapsed from the moment a packet arrives at the MAC layer queue until the packet is successfully transmitted to the intended 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 control mechanism so that the network accepts every new flow. New flows for voice, video, and data traffics periodically arrive to the network every 10 s at the same time. In this figure, Estimated Ratio means the summation of the ratio
measured when k 1 flows contend for the channel and the ratio obtained from (14) for a new kth flow. Allocated Ratio is the usage ratio measured after the new kth flow is admitted. From this figure, we know that the channel time usage ratio almost linearly increases and has a good coincidence under the stable status. At high loads, the ratio results do not match because higher priority traffics take the channel time from lower priority traffics so that the channel time for lower traffics decreases. This is ignorable since a new flow will be blocked by the proposed admission control mechanism when there is no available channel time. This figure demonstrates that the channel time usage ratio obtained from (14) for a new flow is reasonable, and the proposed scheme is able to make right admission decisions. Figs. 5–8 show the performance results when the admission control mechanism is used. The arrival pattern of each traffic is as follows. At the beginning, there are no flows in the network. A new flow periodically arrives to the network every 6 s in the order of 2 s, 4 s, or 6 s for voice, video, or data traffics, respectively. In other words, a voice flow arrives at 2 s, 8 s, 14 s, and so on, a video flow at 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 delayed by a random time chosen from an uniform distribution in the range of [0 s, 0.5 s]. Fig. 5 depicts the channel time usage ratio for the proposed scheme. At light loads, data traffic with higher
1.0 Data Total
0.8
0.6
0.4
0.2
0.0 0
20
40
60
80
100
120
Time (s) Fig. 5. Channel time usage ratio according to the simulation time.
0.5
TxLimit & TxUsed
0.4
Video
0.3
TxLimit Voice 0.2
0.1
TxUsed 0.0 0
20
40
60
80
100
120
Time (s) 0.04
TxLimit TxLimit & TxUsed
Channel Time Usage Ratio
Voice Video
0.03
Video
TxUsed
0.02
Voice 0.01
0.00 0
20
40
60
80
Time (s) Fig. 6. TxLimit and TxUsed according to the simulation time.
100
120
10
S. Kim et al. / Computer Networks 54 (2010) 3–12
14 DAC PAAC
Throughput (Mbps)
12 10 Video
8 6 4
Voice
2 0
0
20
40
60
80
100
120
Time (s)
14 DAC PAAC
Throughput (Mbps)
12 10 8 6 4 2 0 0
20
40
60
80
100
120
Time (s)
Fig. 7. Throughput according to the simulation time.
2.0 DAC PAAC
Delay (ms)
1.5
1.0 Video
0.5
Voice
0.0
0
20
40
60
80
100
120
Time (s) Fig. 8. Delay according to the simulation time.
sending rate uses larger channel time than the other traffics. However, as the number of flows increases, the portion of channel time used by data traffic decreases, since in the proposed scheme higher priority traffics can gain exclusive channel access over lower priority ones. In other
words, to guarantee the QoS requirement for higher priority traffics, the proposed scheme discriminates lower traffics, and reallocates the amount of saved channel time to higher priority traffics. From this figure, we can observe that real-time traffics use as much as the channel time allocated in Table 2. This indicates that the starvation of data traffic can be avoided. Fig. 6 shows TxLimit and TxUsed of the first admitted flows for voice and video traffics of the DAC scheme. In this figure, the unit of Y-axis is the channel time ratio instead of the amount of channel time. For a better readability, we show two figures with a different scale of Y-axis. Once TXOPBudget is depleted, TxLimit converges to TxSuccess SurplusFactor. This ensures that a station can continue consuming the same amount of channel time for subsequent beacon intervals. TxUsed is limited by TxLimit since a station is prevented from sending a packet when TxUsed exceeds TxLimit. This can lead to the performance degradation. As the number of flows becomes larger, collision probability increases so that traffics need more channel time to provide QoS. However, the network cannot provide channel time enough to real-time traffics due to TxLimit based on a constant SurplusFactor. To provide more channel time, SurplusFactor should be larger, thus resulting in fewer admitted flows. From this figure, we can observe that it is not easy to decide a globally optimal SurplusFactor value. Fig. 7 shows the throughput behavior. For the proposed scheme, it shows almost the same patterns as Fig. 5. We see that there are no performance differences between the DAC and PAAC schemes at light loads. For the PAAC scheme, the throughput for voice and video traffics is stable and always met with their requirements regardless of network loads. However, data traffic fluctuates at high loads since voice and video traffics use larger channel time for retransmissions, when they have collided packets. For the DAC scheme, the throughput of video traffic does not remain stable. At high loads, video traffic needs more channel time due to collisions, but it cannot use larger channel time due to TxLimit as shown in Fig. 6. Therefore, the DAC scheme does not meet the throughput requirements, especially from the point of intersection between TxLimit and TxUsed in Fig. 6. Fig. 8 demonstrates the delay performance. We can see that the delay is kept low at light loads. As loads become higher, the proposed scheme outperforms the DAC for all priority traffics. For the PAAC scheme, note that the increase of delay is not due to lower priority traffics. The main reason of voice traffic is that the number of voice flows contending for the channel increases. However, for video traffic, it is because the performance is affected by the number of voice flows as well as its own number of flows. For the DAC scheme, the delay increases sharply. This is from the fact that in the DAC all flows always try to access the channel and make collisions with one another. As illustrated in Figs. 5–8, the proposed scheme totally accepts 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 and maximum channel time for each traffic, the PAAC scheme
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3
Delay (ms)
DAC 2
1
0 0
20
40
60
80
100
120
80
100
120
Time (s) 3
Delay (ms)
PAAC
2
1
0 0
20
40
60
Time (s) Fig. 9. Delay of a voice flow according to traffic loads.
accepts 1 video flow more than the DAC scheme. This indicates that the PAAC scheme is efficient to accommodate more traffic. For the PAAC scheme, after 92 s, no more flows are admitted. When the 13th new data flow arrives at 78 s, about 98% of the channel time is occupied by the admitted flows (i.e., about 20%, 34%, and 44% by voice, video, and data, respectively). Therefore, the new data flow is denied. Similarly, at 92 s and 94 s, voice and video traffics use about 24% of the channel time and about 39% respectively, so that the network cannot admit the 16th new voice and video flows. For the DAC scheme, at 88 s and 92 s, TXOPBudeget is less than the budget required by a new flow for video and voice traffics, respectively. Therefore, the new flows are rejected. Fig. 9 shows the delay performance of a voice flow according to traffic loads. To validate that the proposed scheme can solve the priority reversal problem, we simulated under the environment where the arrival pattern is the same as the one mentioned above except that there is only one voice flow. We increase the channel time for video traffic to accept up to 20 video flows. For the DAC, the delay becomes worse as traffic loads become higher since all priority flows always try to access the channel and make collisions with one another. For the PAAC, the delay remains almost the same and is always met with voice traffic’s requirement under wide range of loads because the proposed scheme ensures that flows of the highest AC always access the channel first. However, at medium loads, it slightly increases compared to light loads. This is from the fact that, in the proposed scheme, voice flows with a new packet arrival during transmission intervals of video or data flows must wait until sensing a packet transmission.
6. Conclusion IEEE 802.11e EDCA does not ensure the QoS requirements of multimedia applications, and causes the performance starvation of lower priority traffics at high loads. In this paper, we proposed a measurement-based admission control scheme to improve the QoS performance. We first propose a priority access mechanism which uses a busy tone. Each priority traffic separately contends for the channel. Therefore, the AP can exactly measure the channel status for each priority traffic. And then the measurement information is used in the admission control mechanism. Simulation results show that the proposed scheme is very effective and ensures the QoS requirements of multimedia applications such as voice and video traffics, and protects the performance starvation of data traffic. References [1] IEEE, Part 11: Wireless LAN Medium Access Control (MAC) and Physical Layer (PHY) Specifications, IEEE Standard 802.11, June 1999. [2] Y. Xiao, A simple and effective priority scheme for IEEE 802.11, IEEE Commun. Lett. 7 (2) (2003) 70–72. [3] H. Zhai, X. Chen, Y. Fang, How well can the IEEE 802.11 wireless LAN support quality of service?, IEEE Trans Wireless Commun. 4 (6) (2005) 3084–3094. [4] IEEE, Part 11: Wireless LAN Medium Access Control (MAC) and Physical Layer (PHY) specifications Amendment: Medium Access Control (MAC) Quality of Service Enhancements, IEEE Standard 802.11e, 2005. [5] D. Deng, R. Chang, A priority scheme for IEEE 802.11 DCF access method, IEICE Trans. Commun. E82-B (1) (1999) 96–102. [6] A. Veres, A.T. Campbell, M. Barry, L. Sun, Supporting service differentiation in wireless packet networks using distributed control, IEEE J. Sel. Areas Commun. 19 (10) (2001) 2081–2093. [7] S. Kim, Y. Cho, QoS enhancement scheme of EDCF in IEEE 802.11e wireless LANs, IEE Electron. Lett. 40 (17) (2004) 1091–1092.
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[8] D. Gao, J. Cai, K. Ngan, Admission control in IEEE 802.11e wireless LANs, IEEE Network 19 (4) (2005) 6–13. [9] D. Pong, T. Moors, Call admission control for IEEE 802.11 contention access mechanism, in: Proc. IEEE GLOBECOM’03, vol. 1, 2003, pp. 174–178. [10] L. Lin, H. Fu, W. Jia, An efficient admission control for IEEE 802.11 networks based on throughput analyses of (un)saturated channel, in: Proc. IEEE GLOBECOM’05, vol. 5, 2005, pp. 3017–3021. [11] J. Zhu, A. Fapojuwo, A new call admission control method for providing desired throughput and delay performance in IEEE802.11e wireless LANs, IEEE Trans. Wireless Commun. 6 (2) (2007) 701–709. [12] X. Chen, H. Zhai, Y. Fang, Supporting QoS in IEEE 802.11e wireless LANs, IEEE Trans. Wireless Commun. 5 (8) (2006) 2217–2227. [13] Y. Xiao, H. Li, S. Choi, Protection and guarantee for voice and video traffic in IEEE 802.11e wireless LANs, in: Proc. IEEE INFOCOM’04, vol. 3, 2004, pp. 2152–2162. [14] Y. Xiao, H. Li, Voice and video transmissions with global data parameter control for the IEEE 802.11e enhance distributed channel access, IEEE Trans. Parallel Distrib. Syst. 15 (11) (2004) 1041–1053. [15] J. Freitag, N. Fonseca, J. Rezende, Admission control in IEEE 802.11 networks, in: Proc. IEEE GLOBECOM’04, 2004, pp. 258–265. [16] X. Yang, N. Vaidya, Priority scheduling in wireless ad hoc networks, Wireless Networks 12 (3) (2006) 273–286. [17] Z.J. Haas, J. Deng, Dual busy tone multiple access (DBTMA)-a multiple access control scheme for ad hoc networks, IEEE Trans. Commun. 50 (6) (2002) 975–985. [18] H. Jiang, P. Wang, W. Zhuang, A distributed channel access scheme with guaranteed priority and enhanced fairness, IEEE Trans. Wireless Commun. 6 (6) (2007) 2114–2125. [19] P. Wang, H. Jiang, W. Zhuang, IEEE 802.11e enhancement for voice service, IEEE Wireless Commun. 13 (1) (2006) 30–35. [20] F. Cali, M. Conti, E. Gregori, IEEE 802.11 protocol: design and performance evaluation of an adaptive backoff mechanism, IEEE J. Sel. Areas Commun. 18 (9) (2000) 1774–1786.
Sunmyeng Kim received the B.S., M.S., and Ph.D. degrees in information and communication from Ajou University, Suwon, Korea, in 2000, 2002, and 2006, respectively. From May 2006 to February 2008, he was a Postdoctoral Researcher in electrical and computer engineering with the University of Florida, Gainesville. In March 2008, he then joined the School of Computer and Software Engineering, Kumoh National Institute of Technology, Gumi, Korea, as a Full-Time Lecturer. His research interests include resource management, wireless LANs and PANs, wireless mesh networks, and quality-ofservice enhancement.
Young-Jong Cho received the B.S. degree in Electronic Engineering from Seoul National University, in 1983 and M.S. and Ph.D. degrees in Electronic Engineering from KAIST in 1985 and 1989, respectively. Since 1996, he has been a professor of the Department of Information and Computer at Ajou University. From 1985 to 1996, he had been a principal engineer at LG Co., Korea. In 1993, he joined the AT&T Bell Lab. as a researcher and in 2003, he was a visiting professor at GMU. His research interests include performance analysis of broadband communication systems and development of high speed Internet devices.
Yong K. Kim received the B.S. degree in Electronics Engineering from Ajou University, Korea in 1988 and the M.S.E. degree in Electrical and Computer Engineering from University of Alabama in Huntsville 1993. He received his Ph.D. degree in Electrical and Computer Engineering in North Carolina State University 2001. He is currently in Associate Professor at School of Electrical Electronics and Information Engineering, Wonkwang University, Korea. His research interests are fiber optics & sensing, fiber property measurement using fiber gratings, photorefractive fiber and crystal devices.
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 b c
School of Computer and Software Engineering, Kumoh National Institute of Technology, Gumi, Republic of Korea Department of Information and Computer Engineering, Ajou University, Suwon, Republic of Korea 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 2008 Received in revised form 7 April 2009 Accepted 9 July 2009 Available online 16 July 2009 Responsible Editor: E. Gregori Keywords: EDCA IEEE 802.11e QoS Admission control Priority access
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 IEEE 802.11e standard. However, it cannot guarantee the stringent real-time constraints of multimedia applications such as voice and video without an efficient method of controlling network 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 measure the channel status per traffic type, we propose a priority access mechanism in which each priority traffic is distinguished by a busy tone, and separately performs its own packet transmission operation. Then, admission control mechanism protects existing flows from new ones, and maintains the QoS of the admitted flows based on the measured channel status information. Performance of the proposed scheme is evaluated by simulation. Our results show that the proposed scheme is very effective in guaranteeing the QoS of multimedia applications as well as in avoiding the performance starvation of low priority traffics. Ó 2009 Elsevier B.V. All rights reserved.
1. Introduction The IEEE 802.11 wireless LAN is widely used for wireless access due to its easy deployment and low cost. The IEEE 802.11 standard defines a medium access control (MAC) protocol for sharing the channel among stations [1]. The distributed coordination function (DCF) was designed for a contention-based channel access. The DCF has two data transmission methods: the default basic access and optional RTS/CTS (request-to-send/clear-to-send) access. The widespread use of multimedia applications requires new features such as high bandwidth and small average delay in wireless LANs. Unfortunately, the IEEE 802.11
q This paper was supported by Research Fund, Kumoh National Institute of Technology. * Corresponding author. Tel.: +82 54 478 7547; fax: +82 54 478 7539. E-mail addresses: [email protected] (S. Kim), [email protected] (Y.-J. Cho), [email protected] (Y.K. Kim).
1389-1286/$ - see front matter Ó 2009 Elsevier B.V. All rights reserved. doi:10.1016/j.comnet.2009.07.002
MAC protocol cannot support quality of service (QoS) requirements [2,3]. In order to support multimedia applications with tight QoS requirements in the IEEE 802.11 MAC protocol, the IEEE 802.11e has been standardized [4]. It introduces a contention-based new channel access mechanism called enhanced distributed channel access (EDCA). The EDCA supports the QoS by introducing four access categories (ACs). To differentiate the ACs, the EDCA uses a set of AC specific parameters, which include minimum contention window, maximum contention window, and arbitration inter-frame space (AIFS). In the previous work [2,5–7], the authors focused on studying the service differentiation. However, service differentiation alone is not sufficient to provide the QoS of multimedia applications without an efficient method of controlling network loads. In this paper, we focus on admission control mechanism for the IEEE 802.11e EDCA. Many admission control mechanisms for 802.11e EDCA have been proposed. The previous mechanisms are classified into two approaches: model-based and
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measurement-based admission control [8]. Model-based admission control approach uses an analytical model to decide whether a new flow is admitted or rejected [9–12]. Measurement-based approach uses measurement information (e.g., throughput, collision probability, or delay) of existing traffic 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 few unpractical hypotheses to calculate the QoS metrics of all existing flows in addition to a new one. They do not reflect the characteristics of real traffics. Therefore, the modelbased mechanisms are always inaccurate and clearly not applicable to realistic environments. Second, it is hard for measurement-based schemes to make a right admission decision since the schemes cannot accurately predict the channel time usage after a new flow is admitted to the network. For example, the scheme proposed in [13,14] allocates the bandwidth to flows and makes an admission decision based on successful transmission time and a constant parameter SurplusFactor. The parameter is used to compensate potential transmission failures because this scheme does not recognize which AC flows send packets when there are collisions. The inaccuracy of this parameter can lead to making a wrong decision. Third, the previous schemes do not solve the priority reversal problem that low priority flows have a shorter backoff counter than higher priority ones, and consequently access the channel [16]. Since each flow decreases its backoff counter when the channel is idle, it will eventually have a small value although its initial counter is large. A high priority flow cannot be always assured to have a smaller backoff counter since the backoff counter is randomly selected based on the uniform distribution. Therefore, lower priority flows can transmit their packets prior to higher priority ones. As a result, higher priority traffic may wait a long time for the channel contention. Fourth, although an admission control scheme works well, higher priority traffics are vulnerable to lower priority traffics since the latter ones may cause many collisions with the former ones. Finally, most of the previous schemes first regulate data traffic in order to provide the QoS to the real-time traffics. Therefore, the performance starvation of data traffic can be caused at high 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, the priority access mechanism uses a short duration busy tone signal (i.e., pulses of energy) [17]. Each flow sends a busy tone in the last slot of its own AIFS period, and then it operates backoff procedure like the EDCA. However, flows that receive the busy tone suspend their busy tone transmission and backoff procedure. This ensures that each priority traffic separately performs its own packet transmission operation. Second, the admission control mechanism regulates network loads and protects admitted flows from new ones. The priority access mechanism is used to facilitate the admission control. The access point (AP) measures the channel status information for each traffic type, and then separately makes an admission decision. The proposed scheme makes the following contributions: (1) The priority access mechanism blocks the trans-
missions of lower priority ACs when higher priority one has packets to transmit. This ensures that flows of the highest AC always access the channel first. Therefore, the proposed scheme can solve the priority reversal problem. (2) The priority access mechanism also ensures the channel contention to be among flows with the same priority AC. In other words, there are no channel contentions and collisions among flows with different priority ACs. Therefore, the AP exactly measures the channel status information for each priority AC, and then accurately predicts the channel time usage after a new flow is admitted. (3) Bandwidth allocation to each priority AC in the previous schemes is extremely challenging due to collisions among different priority ACs. However, the proposed scheme can allocate the bandwidth to each priority AC exactly since there are no collisions among different priority ACs. Accordingly, the proposed scheme can prevent the starvation of data traffic. The paper is organized as follows. In Section 2, we give a brief introduction of the IEEE 802.11e EDCA and the scheme proposed in [13,14]. In Section 3, the proposed priority access mechanism is presented in detail. In Section 4, we describe the admission control mechanism based on the priority access. In Section 5, performance studies are carried out through simulation results. Finally, we draw a conclusion 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 introduces a new medium access method called hybrid coordination function (HCF), which combines a contention-based enhanced distributed access mechanism (EDCA) and a controlled channel access mechanism (HCCA). The EDCA is an enhanced variant of the DCF. In the DCF, all stations contend for the channel with the same priority. On the other hand, the EDCA supports several priority levels by introducing an access category (AC) concept. A station have up to four ACs to support eight user priorities. Each AC is implemented as a separate queue. Each packet arrives at the MAC layer with a priority from higher layer, and is mapped to one AC according to the priority. AC 3, AC 2, AC 1, and AC 0 are for voice, video, best effort data, and background traffics, respectively. In order to differentiate the traffics, the EDCA uses a set of AC specific parameters, which include minimum contention window (CWmin[i]), maximum contention window (CWmax[i]), and arbitration inter-frame space (AIFS[i]) for AC i ði ¼ 0; . . . ; 3Þ. The AIFS is at least distributed inter-frame space (DIFS) long and is calculated with the arbitration inter-frame space number (AIFSN[i]). The duration of AIFS[i] is defined by AIFS½i ¼ SIFS þ AIFSN½i aSlotTime, where SIFS is a short inter-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 that in the above inequalities, at least one must be ‘‘not equal to”. The EDCA assigns a smaller CWmin,CWmax, or AIFS to
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higher priority AC in order to provide the higher chance of accessing the channel. Therefore, in the EDCA, the support of QoS can be achieved by differentiating the probability of accessing 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]. At the first level, the admitted voice and video flows are protected from new and other voice and video flows. The AP measures the amount of channel time (TxTime[i]) occupied by successful transmissions from each AC i during a beacon period. And then, it calculates the additional amount of channel time (TXOPBudget[i]) available for AC i during the next beacon interval.
TXOPBudget½i ¼ maxðATL½i TxTime½i SurplusFactor½i; 0Þ;
ð1Þ
where ATL[i] is the maximum amount of channel time for AC i per beacon interval, and SurplusFactor[i] is used to compensate potential transmission failures due to collisions. The budget information is delivered to each station via a beacon frame. Each station maintains the following variables for AC i: TxUsed[i], TxSuccess[i], TxLimit[i], TxRemainder[i], and TxMemory[i]. TxUsed[i] is the amount of channel time occupied by transmissions, regardless of success or not, from AC i of this station. TxSuccess[i] is the amount of channel time occupied by successful transmissions. A station does not transmit a data packet when TxUsed[i] exceeds TxLimit[i], where how to determine this value is described below. If a station is prevented from sending a frame for this reason, TxRemainder½i ¼ TxLimit½i TxUsed½i. Otherwise, TxRemainder½i ¼ 0. TxMemory[i] memorizes the amount of channel time that AC i of this station utilized during a beacon interval, and is TxMemory½i ¼ a TxMemory½i þ ð1 aÞ ðTxSuccess½i SurplusFactor½iþ TXOPBudget½iÞ. a is a smoothing factor. TxLimit[i] is defined by 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 of AC i, and / is a fraction. If this equation is satisfied, the new flow is admitted. At the second level, the voice and video flows are protected from best-effort data traffic. In order to control data transmissions according to traffic conditions, the DAC scheme dynamically adjusts the EDCA parameters for data traffic. This scheme increases the contention window size much faster than the standard by using a larger contention window increasing factor ri for the backoff stage i instead of 2 used in the standard when a collision occurs. That is ri < rj ðr1 P 2Þ for 1 6 i < j 6 Lretry , where Lretry is the retry limit. Whenever a data frame reaches the retry limit, CWmin and AIFS increase (i.e., CWmin ¼ h CWminðh > 1Þ and AIFS ¼ w AIFSðw > 1Þ). Whenever a station successfully transmits m consecutive packets, CWmin and AIFS decrease (i.e., CWmin ¼ CWmin=hðh > 1Þ and AIFS ¼ AIFS=wðw > 1Þ).
5
3. Priority access In this section we describe the priority access mechanism that ensures the channel contention-based on traffic priority. We consider a single hop IEEE 802.11e network. Our mechanism needs minor modifications to the EDCA. However, the basic operation of the proposed mechanism is the same as that in the EDCA. The proposed mechanism blocks the transmissions of lower priority ACs when higher priority one has packets to transmit. In other words, lower priority ones do not transmit their packets until no higher priority ones contend for the channel. In the EDCA, a flow of a given AC first senses the wireless channel medium. After sensing the idle duration of AIFS period, the flow waits for random backoff time before transmitting. However, in the proposed mechanism, a flow sends a busy tone shorter than aSlotTime after sensing an idle channel for the (AIFS – aSlotTime) period. After sending its own busy tone, the flow operates its packet transmission procedure like the EDCA. So, it decreases its backoff counter as long as the channel is sensed idle, does not decrease when a transmission is detected on the channel, and tries to transmit a packet when the backoff counter reaches zero. On the contrary, if a flow receives a busy tone at any time within its (AIFS – aSlotTime) period, then the flow suspends its busy tone transmission and backoff procedure. In other words, the flow stops its current channel contention, and waits until a packet transmission occurs. As long as at least one flow of higher AC exists, all flows of lower ACs will sense a busy tone within their (AIFS – aSlotTime) periods. For the operation of the proposed mechanism, let us assume that, unlike the EDCA, higher priority AC always has a smaller AIFSN than lower priority one, that is AIFSN½i > AIFSN½j for 0 6 i < j 6 3. The busy tone transmission time varies with the packet arrival time and channel status. In order to determine the time, the proposed mechanism uses three parameters: the packet arrival time (PAT) at the MAC layer, the last channel busy time (LCBT) due to the recent packet transmission, 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 set to the end time of (EIFS DIFS). The reason is that the EDCA backoff operation starts after the channel is determined to be idle for the duration of the (EIFS DIFS + AIFS) period [4], where the EIFS is an extended inter-frame space. There are two possible cases for determining the transmission time of a busy tone (see Fig. 1). If a flow receives a busy tone between LCBT and PAT, it defers its operation until 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 upper layer before LCBT, it sends its busy tone after sensing an idle channel for (AIFS – aSlotTime) period (see Fig. 1a). Second, when a packet arrives after LCBT, the channel should be determined to be idle until (LCBT + LAIFS N + AIFS aSlotTime) before the busy tone is allowed to trans-
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LCBT
New Packet Arrival
AIFS Busy Medium
Backoff Slots
NAV = 0 ?
Time
Y LBT > LCBT ?
Packet arrival
N
Y
N
Busy Tone
PAT > LCBT ?
Slot Time
Y
Defer until LCBT+LAIFS* N
Y
Save LBT, Defer until channel busy
N Listen for AIFS - aSlotTime
LCBT
Rx Busy Tone ?
LAIFS
AIFS
N
Backoff Slots
Busy Medium
Tx Busy Tone
Time Decrease Backoff
Packet arrival Channel idle ?
N
Save LCBT
Y Fig. 1. Busy tone transmission time according to the packet arrival time.
Tx DATA
mit (see Fig. 1b), where N is used to align the start time of AIFS period with the integral multiples of LAIFS, and is PATLCBT . dxe rounds to the smallest integer greater than LAIFS or equal to x. Even though a flow has no packets to transmit, the proposed scheme requires the flow to measure the time passed after the busy medium in the unit of LAIFS so that it can know when to start its own AIFS sense time. This alignment is needed to ensure the channel contention to be among flows with the same priority AC. If a flow sends its busy tone after sensing the idle duration of (AIFS – aSlotTime) period without the alignment, it may cause a busy tone collision with flows of other priority ACs. Then, the flows with the collided busy tone contend for the channel at the same time so that there exist effects among different priority ACs. Before the busy tone transmission time expires, if a busy tone is detected from other flows, then a flow should defer its operation until a packet transmission occurs, and operate as the first case. Fig. 2 shows the operation procedure of the proposed priority access mechanism.
Success ?
Table 1 System parameters. Parameter
Value
Data bit rate Control bit rate Slot Time SIFS Retry limit Propagation delay MAC header FCS PHY PLCP preamble length PHY PLCP header length ACK Beacon interval
54 Mbps 6 Mbps 9 ls 16 ls 7 1 ls 26 Octets 4 Octets 16 ls 5 Octets 14 Octets 200 ms
N
Increase CW
Y Save LCBT, Reset CW to CWmin LAIFS: AIFS of Lowest priority traffic LCBT: Last Channel Busy Time NAV: Network Allocation Vector LBT: Last Busy Tone Rx time CW: Contention Window PAT: Packet Arrival Time N : Alignment Integer Fig. 2. Operation procedure of the priority access mechanism.
In the priority access mechanism, distinguishing a busy tone from a packet transmission is very important to guarantee the proper operation. To do this, the duration of a transmission is used. Transmission time for a packet has at least 3-time slot duration because it includes physical preamble and header of 20 ls, which is from Table 1 in Section 5. A busy tone duration is smaller than one time slot. Estimating the duration is simple without any additional overheads or costs because every station performs the carrier sensing. Each station, by using the carrier sensing, observes the channel status and measures the duration of busy period. Therefore, the proposed mechanism can distinguish between busy tone and packet transmission when receiving a signal. In [18,19], the authors proposed a deterministic priority access scheme, which is similar to our proposed mechanism. However, their scheme is different from ours in the following way. It provides the deterministically prioritized access only to voice traffic over data traffic. A flow sends a busy tone instead of its backoff procedure after sensing an idle channel for AIFS period. The length of the busy tone is equal to its backoff counter. When the flow completes the transmission of the busy tone, it checks the channel status. If the channel is busy, the flow defers the current contention. Otherwise, the flow
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transmits its packets. In this scheme, a packet is transmitted by a flow with the longest busy tone. If there is at least one voice flow, all data flows will sense the busy tone within their AIFS period, and defer their channel contentions.
4. Admission control In this section, we propose an admission control mechanism, which measures some metrics characterizing the wireless channel based on the priority access and then makes an admission decision. The proposed scheme is called priority access-based admission control (PAAC). We consider an IEEE 802.11e BSS (Basic Service Set) with infrastructure. The admission control module is located at the AP of a BSS and is based on the procedure described in the IEEE 802.11e standard [4]. Before a new flow starts its packet transmission, it sends the AP an ADDTS (Add Traffic Stream) Request frame containing its traffic specification in order to request admission. Based on the channel status information measured by the AP and traffic specification information provided by the flow, the admission control module decides whether the flow can be admitted to the network without affecting the QoS of existing flows. If there is no effect, the AP sends an ADDTS Response frame indicating that the request is accepted. Otherwise, a rejection 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 AP needs to accurately predict the channel time usage after a new flow is admitted. An admitted new flow can cause not only extra transmission time, but also more collision time and backoff time. In this subsection, we describe how to obtain the channel time required by a new flow. The AP measures three kinds of channel status information for each AC i ði ¼ 0; . . . ; 3Þ: average channel time usage ratio U i , average collision probability P i , and average backoff time Bi per transmission attempt. The channel time usage ratio is the fraction of time during which the channel is used to transmit each priority traffic. The following two reasons explain why estimating the required information is so simple. First, in the proposed mechanism, each AC separately performs its own packet transmission operation so that unlike the previous measurement-based admission control mechanisms, the proposed mechanism can recognize which AC flows cause collisions. Second, the AP performs the carrier sensing mechanism without any additional overheads or costs. Note that we use the collision probability to compensate transmission failures. We can exactly measure the probability for each traffic priority based on the priority access. In [13,14], the authors allocated the channel time based on successful transmission time and SurplusFactor, since they do not recognize which AC flows cause collisions. Therefore, their scheme may make wrong admission decisions. We assume the channel status information is updated at the end of each beacon interval. For each AC i, the AP ob-
7
serves the channel status and measures the duration of transmission interval TIi , total backoff time BP i , and the number of transmissions NT i and collided transmissions NC i during a beacon interval BI. Transmission interval is the time duration between two consecutive packet transmissions, which is made up of three components: AIFS, backoff, and packet transmission. At the end of a beacon interval, instantaneously measured channel time usage ra, collision probability P measure , and average backtio U measure i i are calculated as off time upon a transmission Bmeasure i follows.
TIi ; BI NC i ¼ ; NT i BPi ¼ : NT i
U measure ¼ i
ð3Þ
P measure i
ð4Þ
Bmeasure i
ð5Þ
From (3)–(5), U i ; Pi , and Bi are approximated by exploiting the following moving averaging window:
U i ¼ aU i þ ð1 aÞU measure ; i
ð6Þ
; P i ¼ aPi þ ð1 aÞP measure i
ð7Þ
; Bi ¼ aBi þ ð1 aÞBmeasure i
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 a value close to 1 makes the reflection of the changed network conditions significantly longer. When the value is small, the fluctuations of the network conditions affect the performance remarkably. Thus in [20], the authors concluded that the choice of 0.9 as smoothing factor is appropriate 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 transmits an ADDTS Request frame with the following parameters: mean packet size Li;k , mean data rate qi;k , and physical transmission rate Ri;k . On receiving an ADDTS Request frame, the AP computes the channel time usage ratio required by the flow. Let N i;k be the number of packets arriving at a new flow k during one beacon interval,
Ni;k ¼
qi;k BI Li;k
:
ð9Þ
When there are collisions during transmissions, the flow retransmits its collided packets. Therefore, required channel time should include the packets to be retransmitted. Total number of packets to be transmitted is given by
Ni;k;total ¼
Ni;k : 1 Pi
ð10Þ
The channel time required to transmit N i;k;total packets is as follows:
ChTimei;k ¼ Ni;k T s þ ðNi;k;total Ni;k ÞT c þ N i;k;total ðBi þ AIFS½iÞ; ð11Þ where T c and T s are the average time intervals during which the channel is sensed busy due to collision and successful transmission, respectively. For the basic and RTS/ CTS access methods, T c and T s are given by
S. Kim et al. / Computer Networks 54 (2010) 3–12
T c;basic ¼ H þ L þ EIFS DIFS;
ð12Þ
T s;basic ¼ H þ L þ SIFS þ ACK þ 2d; T c;rts=cts ¼ RTS þ EIFS DIFS;
T s;rts=cts ¼ RTS þ SIFS þ CTS þ SIFS þ H þ L þ SIFS þ ACK þ 4d; ð13Þ where Hð¼ PHYhdr þ MAChdrÞ is the time to transmit a packet header, d is propagation delay, SIFS is the SIFS time space, RTS,CTS, and ACK are the times to transmit RTS, CTS, and ACK frames, respectively. L is the packet transmission time, which is the ratio of Li;k and Ri;k . The channel time usage ratio required by the new flow is given by
U i;k;require ¼
ChTimei;k : BI
ð14Þ
4.2. Admission decision In the priority access mechanism, each priority traffic separately performs its own packet transmission operation. In order to make admission decisions according to the traffic type, the channel time is allocated to each traffic type (see Fig. 3). Let C be the total channel time in which f1 C portion is reserved for voice traffic, f3 C portion is for video traffic, f2 C portion is for sharing between voice and video traffics, and f4 C portion is for data traffic ðf1 þ f2 þ f3 þ f4 ¼ 1Þ. These portions could be changed based on the traffic status among voice, video, and data. Previous schemes cannot allocate the channel time to each traffic exactly due to collisions among different AC traffics. However, the proposed scheme can do it since collisions among different AC traffics can be eliminated by the priority access mechanism. Partitioning the channel time ensures that voice or/and video traffics do not monopolize the channel time, and data traffic can avoid performance starvation. Data traffic uses the channel time unused by voice or/and video traffics for high channel utilization. On receiving a new ADDTS Request frame, the AP makes a decision whether or not to accept the flow. For a new flow k of AC i, we have the following decision criterion:
U i þ U i;k;require 6 CT i ;
ð15Þ
where CT i is the channel time usage ratio which can be used for AC i at the moment when the new request is received. This value varies dynamically with time since voice and video traffics share f2 of the channel time, and data traffic uses the channel time unused by voice or/and video traffics. CT i is ðf1 þ f2 þ f3 maxðU video ; f3 ÞÞ for voice traffic, ðf1 þ f2 þ f3 maxðU voice ; f1 ÞÞ for video traffic, and ð1 U voice U video Þ for data traffic. If (15) is satisfied, the new flow request is accepted. Otherwise, it is denied. Total Channel Time (C) f1 C
f2 C
f3 C
f4 C
Voice
Voice/Video
Video
Data
5. Performance evaluation In this section, we discuss the performance of the proposed scheme. We have implemented the proposed scheme with the NS-2 simulator. System parameters used in the simulation are listed in Table 1. We simulated an IEEE 802.11a network with transmission rates of 54 Mbps for data packets and of 6 Mbps for control packets, respectively. We have three types of traffics: voice, video, and data. Traffic parameters are listed in Table 2. A constant bit rate (CBR) model is used for all three traffics. We allocate 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 ¼ ATL½1 ¼ 0:35). f2 is not allocated for performance comparison with the DAC scheme. Data traffic is allowed to use the channel time unused by real-time traffics. In the proposed scheme, the number of flows contending with each other at the same time is fewer than the standard MAC since each priority traffic separately contends for the channel, so that CWmin and CWmax can be set to smaller values. For the proposed scheme, we set (CWmin, CWmax) to (3, 7), (7, 15), and (15, 511) for voice, video, and data traffics, respectively. For the DAC scheme, we have SurplusFactor ¼ 1:2; and riþ1 ¼ 2ri ði ¼ 1; . . . ; / ¼ 1; m ¼ 1; u ¼ 1; h ¼ 1:5, Lretry 1Þ [13]. In the simulation, we consider a single hop wireless LAN with basic access method and only uplink traffics. We assume that each station has a single flow of voice, video, or data traffic, respectively. The simulation time is 120 s.
Table 2 Traffic parameters. Parameter
Voice
Video
Data
AIFSN CWmin CWmax Packet size (Octets) Inter arrival time (ms) Sending rate (Kbps) Channel time (%)
2 7 15 120 10 96 25
4 15 31 1000 12.5 640 40
7 31 1023 1500 12.5 960 35
1.0
Channel Time Usage Ratio
8
Estimated Ratio Allocated Ratio
0.8
Video
0.6 Data
0.4 Voice
0.2
0.0 0
5
10
15
20
25
Number of Flows per Traffic Fig. 3. Channel time allocation for each traffic type.
Fig. 4. Channel time usage ratio required for a new flow.
30
9
S. Kim et al. / Computer Networks 54 (2010) 3–12
Main performance metrics of interest are throughput, delay, and channel time usage ratio. Delay is the time elapsed from the moment a packet arrives at the MAC layer queue until the packet is successfully transmitted to the intended 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 control mechanism so that the network accepts every new flow. New flows for voice, video, and data traffics periodically arrive to the network every 10 s at the same time. In this figure, Estimated Ratio means the summation of the ratio
measured when k 1 flows contend for the channel and the ratio obtained from (14) for a new kth flow. Allocated Ratio is the usage ratio measured after the new kth flow is admitted. From this figure, we know that the channel time usage ratio almost linearly increases and has a good coincidence under the stable status. At high loads, the ratio results do not match because higher priority traffics take the channel time from lower priority traffics so that the channel time for lower traffics decreases. This is ignorable since a new flow will be blocked by the proposed admission control mechanism when there is no available channel time. This figure demonstrates that the channel time usage ratio obtained from (14) for a new flow is reasonable, and the proposed scheme is able to make right admission decisions. Figs. 5–8 show the performance results when the admission control mechanism is used. The arrival pattern of each traffic is as follows. At the beginning, there are no flows in the network. A new flow periodically arrives to the network every 6 s in the order of 2 s, 4 s, or 6 s for voice, video, or data traffics, respectively. In other words, a voice flow arrives at 2 s, 8 s, 14 s, and so on, a video flow at 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 delayed by a random time chosen from an uniform distribution in the range of [0 s, 0.5 s]. Fig. 5 depicts the channel time usage ratio for the proposed scheme. At light loads, data traffic with higher
1.0 Data Total
0.8
0.6
0.4
0.2
0.0 0
20
40
60
80
100
120
Time (s) Fig. 5. Channel time usage ratio according to the simulation time.
0.5
TxLimit & TxUsed
0.4
Video
0.3
TxLimit Voice 0.2
0.1
TxUsed 0.0 0
20
40
60
80
100
120
Time (s) 0.04
TxLimit TxLimit & TxUsed
Channel Time Usage Ratio
Voice Video
0.03
Video
TxUsed
0.02
Voice 0.01
0.00 0
20
40
60
80
Time (s) Fig. 6. TxLimit and TxUsed according to the simulation time.
100
120
10
S. Kim et al. / Computer Networks 54 (2010) 3–12
14 DAC PAAC
Throughput (Mbps)
12 10 Video
8 6 4
Voice
2 0
0
20
40
60
80
100
120
Time (s)
14 DAC PAAC
Throughput (Mbps)
12 10 8 6 4 2 0 0
20
40
60
80
100
120
Time (s)
Fig. 7. Throughput according to the simulation time.
2.0 DAC PAAC
Delay (ms)
1.5
1.0 Video
0.5
Voice
0.0
0
20
40
60
80
100
120
Time (s) Fig. 8. Delay according to the simulation time.
sending rate uses larger channel time than the other traffics. However, as the number of flows increases, the portion of channel time used by data traffic decreases, since in the proposed scheme higher priority traffics can gain exclusive channel access over lower priority ones. In other
words, to guarantee the QoS requirement for higher priority traffics, the proposed scheme discriminates lower traffics, and reallocates the amount of saved channel time to higher priority traffics. From this figure, we can observe that real-time traffics use as much as the channel time allocated in Table 2. This indicates that the starvation of data traffic can be avoided. Fig. 6 shows TxLimit and TxUsed of the first admitted flows for voice and video traffics of the DAC scheme. In this figure, the unit of Y-axis is the channel time ratio instead of the amount of channel time. For a better readability, we show two figures with a different scale of Y-axis. Once TXOPBudget is depleted, TxLimit converges to TxSuccess SurplusFactor. This ensures that a station can continue consuming the same amount of channel time for subsequent beacon intervals. TxUsed is limited by TxLimit since a station is prevented from sending a packet when TxUsed exceeds TxLimit. This can lead to the performance degradation. As the number of flows becomes larger, collision probability increases so that traffics need more channel time to provide QoS. However, the network cannot provide channel time enough to real-time traffics due to TxLimit based on a constant SurplusFactor. To provide more channel time, SurplusFactor should be larger, thus resulting in fewer admitted flows. From this figure, we can observe that it is not easy to decide a globally optimal SurplusFactor value. Fig. 7 shows the throughput behavior. For the proposed scheme, it shows almost the same patterns as Fig. 5. We see that there are no performance differences between the DAC and PAAC schemes at light loads. For the PAAC scheme, the throughput for voice and video traffics is stable and always met with their requirements regardless of network loads. However, data traffic fluctuates at high loads since voice and video traffics use larger channel time for retransmissions, when they have collided packets. For the DAC scheme, the throughput of video traffic does not remain stable. At high loads, video traffic needs more channel time due to collisions, but it cannot use larger channel time due to TxLimit as shown in Fig. 6. Therefore, the DAC scheme does not meet the throughput requirements, especially from the point of intersection between TxLimit and TxUsed in Fig. 6. Fig. 8 demonstrates the delay performance. We can see that the delay is kept low at light loads. As loads become higher, the proposed scheme outperforms the DAC for all priority traffics. For the PAAC scheme, note that the increase of delay is not due to lower priority traffics. The main reason of voice traffic is that the number of voice flows contending for the channel increases. However, for video traffic, it is because the performance is affected by the number of voice flows as well as its own number of flows. For the DAC scheme, the delay increases sharply. This is from the fact that in the DAC all flows always try to access the channel and make collisions with one another. As illustrated in Figs. 5–8, the proposed scheme totally accepts 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 and maximum channel time for each traffic, the PAAC scheme
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S. Kim et al. / Computer Networks 54 (2010) 3–12
3
Delay (ms)
DAC 2
1
0 0
20
40
60
80
100
120
80
100
120
Time (s) 3
Delay (ms)
PAAC
2
1
0 0
20
40
60
Time (s) Fig. 9. Delay of a voice flow according to traffic loads.
accepts 1 video flow more than the DAC scheme. This indicates that the PAAC scheme is efficient to accommodate more traffic. For the PAAC scheme, after 92 s, no more flows are admitted. When the 13th new data flow arrives at 78 s, about 98% of the channel time is occupied by the admitted flows (i.e., about 20%, 34%, and 44% by voice, video, and data, respectively). Therefore, the new data flow is denied. Similarly, at 92 s and 94 s, voice and video traffics use about 24% of the channel time and about 39% respectively, so that the network cannot admit the 16th new voice and video flows. For the DAC scheme, at 88 s and 92 s, TXOPBudeget is less than the budget required by a new flow for video and voice traffics, respectively. Therefore, the new flows are rejected. Fig. 9 shows the delay performance of a voice flow according to traffic loads. To validate that the proposed scheme can solve the priority reversal problem, we simulated under the environment where the arrival pattern is the same as the one mentioned above except that there is only one voice flow. We increase the channel time for video traffic to accept up to 20 video flows. For the DAC, the delay becomes worse as traffic loads become higher since all priority flows always try to access the channel and make collisions with one another. For the PAAC, the delay remains almost the same and is always met with voice traffic’s requirement under wide range of loads because the proposed scheme ensures that flows of the highest AC always access the channel first. However, at medium loads, it slightly increases compared to light loads. This is from the fact that, in the proposed scheme, voice flows with a new packet arrival during transmission intervals of video or data flows must wait until sensing a packet transmission.
6. Conclusion IEEE 802.11e EDCA does not ensure the QoS requirements of multimedia applications, and causes the performance starvation of lower priority traffics at high loads. In this paper, we proposed a measurement-based admission control scheme to improve the QoS performance. We first propose a priority access mechanism which uses a busy tone. Each priority traffic separately contends for the channel. Therefore, the AP can exactly measure the channel status for each priority traffic. And then the measurement information is used in the admission control mechanism. Simulation results show that the proposed scheme is very effective and ensures the QoS requirements of multimedia applications such as voice and video traffics, and protects the performance starvation of data traffic. References [1] IEEE, Part 11: Wireless LAN Medium Access Control (MAC) and Physical Layer (PHY) Specifications, IEEE Standard 802.11, June 1999. [2] Y. Xiao, A simple and effective priority scheme for IEEE 802.11, IEEE Commun. Lett. 7 (2) (2003) 70–72. [3] H. Zhai, X. Chen, Y. Fang, How well can the IEEE 802.11 wireless LAN support quality of service?, IEEE Trans Wireless Commun. 4 (6) (2005) 3084–3094. [4] IEEE, Part 11: Wireless LAN Medium Access Control (MAC) and Physical Layer (PHY) specifications Amendment: Medium Access Control (MAC) Quality of Service Enhancements, IEEE Standard 802.11e, 2005. [5] D. Deng, R. Chang, A priority scheme for IEEE 802.11 DCF access method, IEICE Trans. Commun. E82-B (1) (1999) 96–102. [6] A. Veres, A.T. Campbell, M. Barry, L. Sun, Supporting service differentiation in wireless packet networks using distributed control, IEEE J. Sel. Areas Commun. 19 (10) (2001) 2081–2093. [7] S. Kim, Y. Cho, QoS enhancement scheme of EDCF in IEEE 802.11e wireless LANs, IEE Electron. Lett. 40 (17) (2004) 1091–1092.
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[8] D. Gao, J. Cai, K. Ngan, Admission control in IEEE 802.11e wireless LANs, IEEE Network 19 (4) (2005) 6–13. [9] D. Pong, T. Moors, Call admission control for IEEE 802.11 contention access mechanism, in: Proc. IEEE GLOBECOM’03, vol. 1, 2003, pp. 174–178. [10] L. Lin, H. Fu, W. Jia, An efficient admission control for IEEE 802.11 networks based on throughput analyses of (un)saturated channel, in: Proc. IEEE GLOBECOM’05, vol. 5, 2005, pp. 3017–3021. [11] J. Zhu, A. Fapojuwo, A new call admission control method for providing desired throughput and delay performance in IEEE802.11e wireless LANs, IEEE Trans. Wireless Commun. 6 (2) (2007) 701–709. [12] X. Chen, H. Zhai, Y. Fang, Supporting QoS in IEEE 802.11e wireless LANs, IEEE Trans. Wireless Commun. 5 (8) (2006) 2217–2227. [13] Y. Xiao, H. Li, S. Choi, Protection and guarantee for voice and video traffic in IEEE 802.11e wireless LANs, in: Proc. IEEE INFOCOM’04, vol. 3, 2004, pp. 2152–2162. [14] Y. Xiao, H. Li, Voice and video transmissions with global data parameter control for the IEEE 802.11e enhance distributed channel access, IEEE Trans. Parallel Distrib. Syst. 15 (11) (2004) 1041–1053. [15] J. Freitag, N. Fonseca, J. Rezende, Admission control in IEEE 802.11 networks, in: Proc. IEEE GLOBECOM’04, 2004, pp. 258–265. [16] X. Yang, N. Vaidya, Priority scheduling in wireless ad hoc networks, Wireless Networks 12 (3) (2006) 273–286. [17] Z.J. Haas, J. Deng, Dual busy tone multiple access (DBTMA)-a multiple access control scheme for ad hoc networks, IEEE Trans. Commun. 50 (6) (2002) 975–985. [18] H. Jiang, P. Wang, W. Zhuang, A distributed channel access scheme with guaranteed priority and enhanced fairness, IEEE Trans. Wireless Commun. 6 (6) (2007) 2114–2125. [19] P. Wang, H. Jiang, W. Zhuang, IEEE 802.11e enhancement for voice service, IEEE Wireless Commun. 13 (1) (2006) 30–35. [20] F. Cali, M. Conti, E. Gregori, IEEE 802.11 protocol: design and performance evaluation of an adaptive backoff mechanism, IEEE J. Sel. Areas Commun. 18 (9) (2000) 1774–1786.
Sunmyeng Kim received the B.S., M.S., and Ph.D. degrees in information and communication from Ajou University, Suwon, Korea, in 2000, 2002, and 2006, respectively. From May 2006 to February 2008, he was a Postdoctoral Researcher in electrical and computer engineering with the University of Florida, Gainesville. In March 2008, he then joined the School of Computer and Software Engineering, Kumoh National Institute of Technology, Gumi, Korea, as a Full-Time Lecturer. His research interests include resource management, wireless LANs and PANs, wireless mesh networks, and quality-ofservice enhancement.
Young-Jong Cho received the B.S. degree in Electronic Engineering from Seoul National University, in 1983 and M.S. and Ph.D. degrees in Electronic Engineering from KAIST in 1985 and 1989, respectively. Since 1996, he has been a professor of the Department of Information and Computer at Ajou University. From 1985 to 1996, he had been a principal engineer at LG Co., Korea. In 1993, he joined the AT&T Bell Lab. as a researcher and in 2003, he was a visiting professor at GMU. His research interests include performance analysis of broadband communication systems and development of high speed Internet devices.
Yong K. Kim received the B.S. degree in Electronics Engineering from Ajou University, Korea in 1988 and the M.S.E. degree in Electrical and Computer Engineering from University of Alabama in Huntsville 1993. He received his Ph.D. degree in Electrical and Computer Engineering in North Carolina State University 2001. He is currently in Associate Professor at School of Electrical Electronics and Information Engineering, Wonkwang University, Korea. His research interests are fiber optics & sensing, fiber property measurement using fiber gratings, photorefractive fiber and crystal devices.