ivoip: an Intelligent Bandwidth Management Scheme for VoIP in WLANs

Transcription

1 VoIP: an Intellgent Bandwdth Management Scheme for VoIP n WLANs Zhenhu Yuan and Gabrel-Mro Muntean Abstract Voce over Internet Protocol (VoIP) has been wdely used by many moble consumer devces n IEEE wreless local area networks (WLAN) due to ts low cost and convenence. However, delays of all VoIP flows dramatcally ncrease when network capacty s approached. Addtonally, unfar traffc dstrbuton between downlnk and uplnk flows n WLANs mpacts the perceved VoIP qualty. Ths paper proposes an ntellgent bandwdth management scheme for VoIP servces (VoIP) that mproves bandwdth utlzaton and provdes far downlnk-uplnk channel access. VoIP s a cross-layer soluton whch ncludes two components: 1) VoIP-Admsson Control, whch protects the qualty of exstng flows and ncreases the utlzaton of wreless network resources; 2) VoIP-Farness scheme, whch balances the channel access opportunty between access pont (AP) and wreless statons. VoIP-Admsson Control lmts the number of VoIP flows based on an estmaton of VoIP capacty. VoIP-Farness mplements a contenton wndow adaptaton scheme at AP whch uses stereotypes and consders several major QoS parameters to balance the network access of downlnk and uplnk flows, respectvely. Extensve smulatons and real tests have been performed, demonstratng that VoIP has both very good VoIP capacty estmaton and admsson control results. Addtonally, VoIP mproves the downlnk/uplnk farness level n terms of throughput, delay, loss, and VoIP qualty. Index Terms VoIP, QoS, admsson control, farness, downlnk/uplnk, IEEE I. INTRODUCTION IEEE wreless local area networks (WLAN) have been wdely deployed as the last-mle Internet access n homes, unverstes and enterprses [1]. Along wth the sgnfcant growth of WLAN connectons, the popularty of multmeda delvery servces s also ncreasng [2], ncludng web-browsng, e-mals, on-lne games, vdeo streamng, and socal networkng. Meanwhle, popular VoIP software products, such as Skype 1 and Google Talk 2, have been supported by the majorty of moble consumer devces and have attracted mllons of users [3]. An nterestng nvestgaton shows that more than 50 percent of voce calls orgnate from ndoor WLANs [4]. Fg.1 shows how multple moble devces can be connected to WLANs va access ponts (APs). Moble VoIP users n dfferent WLANs communcate through a remote VoIP server. Unfortunately, the orgnal protocol does not support Ths work was supported n part by Enterprse Ireland Innovaton Partnershp programme. Dr. Z.Yuan and Dr. G.-M.Muntean are wth the Performance Engneerng Lab, School of Electronc Engneerng, Dubln Cty Unversty, Ireland (e-mal: zhenhu.yuan2@mal.dcu.e and munteang@eeng.dcu.e). 1 Skype, Avalable [Onlne]

2 Internet Qualty of Servce (QoS) provsonng for real tme applcatons such as VoIP. Consequently, many solutons have been proposed to provde QoS support for multmeda servces and some focus on VoIP applcatons n wreless networks [5], [6], [7]. Admsson control and downlnk/uplnk farness are two crtcal ssues when delverng VoIP servces over IEEE networks. Frst, t s mperatve to have an effcent admsson control mechansm for VoIP servces n order to provde a consstent level of QoS, especally n terms of delay, to exstng traffc. Most of the current admsson control technques are based on ether bandwdth estmaton [8], [9], [10], [11] or applcaton level QoS requrements [12], [13], [14]. Secondly, the QoS level of VoIP servces s also affected by the unfar traffc dstrbuton between downlnk and uplnk n the IEEE WLAN [15]. In orgnal protocols n nfrastructure mode, the downlnk flows obtan less channel access opportunty than the uplnk flows, as AP has only one MAC queue and one backoff access entty. IEEE e [16] standard updates the MAC layer of the orgnal standard by usng four access categores (ACs) that use separate queues and backoff nstances. However, multple downlnk flows belongng to an AC of the AP have the same prorty wth the uplnk flows belongng to the same AC of the statons. Therefore, the downlnk/uplnk unfarness problem mght stll be caused by the flows from the same AC. Consequently, there s a need for a far traffc dstrbuton between the AP (at the core of the traffc management) and wreless statons. VoIP Clent VoIP Clent VoIP Clent AP company IEEE WLAN Gateway VoIP Server VoIP Provder VoIP Clent AP home VoIP Clent VoIP Clent IEEE WLAN Fg. 1 Scenaro of IEEE based VoIP applcatons 2 Google Talk, Avalable [Onlne]

3 Many farness-orented solutons have been developed focusng on modfyng the IEEE MAC settngs ncludng Contenton Wndows (CW) [17], [18], [19], Transmsson Opportunty (TXOP) [20], and Arbtraton Inter Frame Space (AIFS) [21], [22]. However, these farness-based solutons do not take nto account the QoS level of both downlnk and uplnk traffc and do not consder upper layer parameters. For nstance, n order to farly balance QoS levels, t s necessary to make use of the values of several QoS-related parameters n conjuncton such as throughput, delay, and loss, snce they are all crtcal for the VoIP traffc. Addtonally, none of the prevous research works provde both admsson control and downlnk/uplnk farness support for VoIP servces n IEEE networks. Ths paper proposes a novel ntellgent Bandwdth Management soluton (VoIP) for VoIP servces n IEEE WLANs. VoIP guarantees desred QoS levels by ntroducng a new bandwdth estmaton-based admsson control mechansm. Addtonally, a novel contenton wndow (CW) adaptaton scheme s desgned to acheve QoS farness between downlnk and uplnk. Ths scheme reles on a stereotypes-based algorthm, whch utlzes the rato between the major QoS parameter values (.e. throughput, delay, and loss) measured for downlnk and uplnk traffc, respectvely. Stereotypes for managng groups of users were frst ntroduced by Rch n the Grundy system [23] and they are stll wdely used by many QoS-orented adaptve solutons [24], [25]. Ths paper s structured as follows. Secton II dscusses the related works on admsson control and farness schemes n WLANs. Secton III presents the prncple of VoIP and the archtecture of the VoIP system. Secton IV and secton V ntroduce the expermental setup and present results analyss, respectvely. Conclusons are drawn n secton VI. II. RELATED WORKS Ths secton brefly summarzes related works regardng admsson control and downlnk/uplnk farness technques n wreless networks. A. Admsson Control Technques The purpose of admsson control n wreless networks s to restrct the traffc load n order to mantan QoS at certan level for the exstng flows. The core mechansm of admsson control s the admsson decson polcy. An admsson control algorthm based on the IEEE e EDCA [16] protocol s proposed n [9] by takng nto account the dynamc network condtons. A new flow s admtted only f t wll not cause the overall admtted flows used bandwdth to exceed the wreless network capacty, whch s estmated based on a MAC analytcal model [26]. Another admsson control scheme, Traffc Stream-Admsson Control (TS-AC) [10], was developed to maxmze the wreless network utlzaton. When a

4 new VoIP call request comes, TS-AC frst explots the flow s characterstcs and then compares the flow s requested bandwdth to the measured unused bandwdth. The call s admtted f the comparson s outcome s postve, otherwse, t s rejected. In [11], a novel call admsson control scheme wth a pollng-based schedulng polcy for CBR traffc s proposed n IEEE e wrelelss LAN. The proposed transmt-permsson polcy for HCF controlled channel access (HCCA) protocol can predct the maxmum delay for each packet and derve suffcent condtons so that all the CBR sources satsfy ther tme constrants to ensure QoS levels. Smulatons are conducted and results show that the proposed scheme provdes a hgh throughput wth respect to the system load. In [12], a new IEEE e flow-based admsson control scheme s developed to control and adjust channel access parameters wth the channel condton varatons. Each wreless staton classfes the arrvng request and calculates the admsson control parameters for the flow based on ts maxmum tolerable collson rate (CR max ). When CR max s less than the current channel collson rate, the requred delay and droppng rate of ths flow cannot be satsfed under the current channel condtons, and therefore, the flow request should be rejected. In [13], a novel admsson control scheme s proposed to mprove VoIP qualty n multservce wreless cellular networks. It consders delays as a resource shared by all system users and regularly measures VoIP packet delays. The admsson control scheme verfes f the current VoIP delays added to the estmated resource demand of the ncomng sesson are hgher or lower than the admsson threshold for that type of servce. If hgher, the access of the ncomng sesson s blocked, otherwse, t s allowed. A recent call admsson control scheme [14] probes the network to determne f a VoIP flow can be supported wth acceptable QoS. The probng procedure utlzes Internet Control Message Protocol (ICMP) echo messages (png) to measure Round-Trp Tme (RTT), jtter, and packet loss. The call s then admtted f the measured RTT, jtter, and loss do not exceed predetermned admsson thresholds. In concluson, performance of admsson control schemes for wreless networks largely depends on two factors: 1) accurate analyss of the wreless network condtons; 2) QoS levels requred by applcatons. B. Downlnk/Uplnk Farness Technques Farness ssue between downlnk and uplnk flows refers to far channel access between AP and wreless statons. Current farness-orented schemes n wreless networks focus on adaptng MAC parameters,.e. mnmum or maxmum CW, TXOP, AIFS, etc. In [17], farness between downlnk and uplnk flows n IEEE networks s consdered, where uplnk flows domnate over downlnk flows n terms of channel usage. The proposed scheme dynamcally controls the mnmum sze of contenton wndow (CW mn ) at AP accordng to a computed optmal rato between the packet transmsson rate of downlnk and uplnk flows. Packet rate s defned as a functon of the numbers of uplnk and downlnk flows, as well as the mnmum contenton wndow szes at the AP and wreless termnals. [17] frstly provdes a smplfed analyss of the packet rate rato R, whch does not strongly depend on

5 the number of uplnk flows (N U ). Ths s then supported by the mean feld approxmaton analyss. Secondly, an explct formula for the optmal CW mn at the AP was derved, assumng N U = 1, and usng t as a quas-optmal CW mn at the AP for all N U 1. In [19], VoIP capacty s ncreased through adaptve frame aggregaton and downlnk/uplnk bandwdth far dstrbuton. In partcular, the soluton uses mnmum contenton wndow (CW mn ) adaptaton, whch can precsely control the bandwdth dstrbuton among wreless statons. The bandwdth share rato s nversely proportonal to CW mn rato. The CW mn for the AP s set to the 1/(k-1) of that for actvely transmttng wreless statons. Dfferent from exstng TXOP-based farness scheme, the soluton n [20] dynamcally controls the contenton wndow and TXOP accordng to the packet error rate and the number of statons. In [20], farness levels between downlnk and uplnk flows are mproved by controllng both TXOP lmt and CW mn sze. The prncple of the scheme reles on the TXOP dfferentaton approach, whch allows AP to send multple frames per one channel access. In error-free envronments, farness between downlnk and uplnk s acheved by settng AP s TXOP lmt to the tme requred for n d (the number of downlnk statons) frame transmssons. When channel error s large, ncreasng AP s TXOP lmt s not effcent. Therefore, AP sets ts TXOP lmt for transmttng n d frames, and dynamcally adjusts CW mn accordng to channel error rate. [21] proposes a dstrbuted soluton to control the channel usage of each statons n IEEE e networks, where the AIFS s prortzed for dfferent servce classes. The prncple s to modulate the dstrbuton of staton s AIFS and analyze the threshold of AIFS that can acheve far channel access between downlnk and uplnk. [22] mproves TCP farness n e WLANs n terms of asymmetry between TCP downlnk and uplnk flows. IEEE e AIFS and CW mn parameters are used to assgn hgher prorty to AP. A small value of AIFS and CW mn results n near strct prortzaton of TCP ACKs at the AP. A larger value of CW mn s set at each wreless staton n order to reduce contenton between competng TCP ACKs. Addtonally, e TXOP mechansm provdes a fne graned way for prortzng TCP downlnk data packets. IEEE e EDCA mproves DCF by usng access categores to support prortes between dfferent traffc classes. Most current APs adopt EDCA snce t s dstrbuted and easer to mplement. The dea of TXOP s ntroduced by the IEEE e HCF Controlled Channel Access (HCCA) protocol. HCCA allows wreless statons to send multple contenton-free packets and assgns hgher channel access opportunty n comparson wth EDCA. Nevertheless, HCCA s not wdely deployed n ad-hoc networks due to ts centralzed approach. However, to the best of our knowledge, none of the prevous research works consder farness between downlnk and uplnk traffc n networks n terms of QoS levels, as proposed n ths paper. III. IVOIP SYSTEM ARCHITECTURE

6 VoIP performs admsson control based on bandwdth avalablty and mproves farness accordng to QoS levels between downlnk and uplnk traffc. Fg. 2 llustrates the TCP/IP protocol stack model-based archtecture of the VoIP system. It conssts of three components: 1) Model-based Bandwdth Estmaton (MBE) [27] [28], whch s located at applcaton layer and estmates the avalable bandwdth resources of IEEE network; 2) VoIP-Admsson Control, whch s located at Fg. 2 Block archtecture of VoIP system Start Flow requests jonng the WLAN VoIP-Admsson Control Reject flow No Decde whether accept flow Yes VoIP-Farness calculate CW for End flow & Delver flow packets Fg. 3 System algorthm of VoIP

7 applcaton layer and lmts the amount of VoIP traffc admtted nto an IEEE network; 3) VoIP-Farness, whch s deployed at MAC layer and balances the downlnk/uplnk channel access opportunty. VoIP-Admsson Control module receves feedback (.e. avalable bandwdth) from MBE and VoIP-Farness module receves nformaton (.e. QoS levels) from the MAC layer. VoIP traffc s delvered usng the Real-tme Transport Protocol (RTP) [29] and feedback nformaton s collected and delvered usng the mechansm provded by IEEE [30] framework. Detals of each major component of the VoIP system are ntroduced next. The decson-makng procedure of VoIP s shown n Fg. 3. When a new VoIP flow requests jonng the wreless LAN, the VoIP-Admsson Control module estmates the overall avalable bandwdth and decdes whether to admt or not the new flow n order to avod network overload and congeston. If flow s admtted to the wreless network, then VoIP-Farness module s used to assgn proper channel access opportunty to flow, and n meanwhle, provde far access between exstng downlnk and uplnk flows. Ths s done by usng the stereotypes-based resource allocaton and adjustng the contenton wndow sze of flow. Detaled descrpton of the system modules wll be provded n the followng sectons. A. Model-based Bandwdth Estmaton (MBE) MBE [28] estmates the avalable bandwdth based on novel TCP and throughput models for IEEE WLANs. MBE s an applcaton layer soluton and uses an IEEE Meda Independent Handhover (MIH) Functon module to montor the transmsson-related nformaton such as packet loss, round trp delay, and packet sze. MBE estmates the bandwdth n three steps as follows. Frst, MBE combnes the TCP throughput model [31] and the IEEE DCF [32] model, n order to consder both TCP congeston control and wreless characterstcs. The bandwdth acheved by a TCP flow (B TCP ) s gven n equaton (1), where b s the number of packets acknowledged by a receved ACK, TCP P retr s the probablty of packet retransmsson, MRTT s the transport layer round-trp tme between sender and recever, and MSS denotes the maxmum segment sze. T o s the tmeout value to trgger retransmsson. B TCP MRTT 2 TCP TCP retr retr TCP TCP T mn(1,3 ) ( o Pretr Pretr bp 3 MSS 3bP 8 ) (1) Unlke TCP, the protocol does not support packet retransmssons and therefore the over WLAN throughput model should consder ths. Hence, the terms P retr and MRTT defned n equatons (1) whch consder TCP fast retransmsson and tmeout respectvely should be removed n MBE s verson. Accordng to [28], the probabltes of retransmsson ( P ) retr

8 and successful transmsson when traffc runs over IEEE networks can be gven as shown n equaton (2) and (3), respectvely, where P DCF s the transmsson loss probablty and P drop refers to the drop probablty: P retr P DCF P drop (2) P succ 1 P (3) retr The average delay, Delay_, for successfully transmttng the ndvdual packet could be wrtten as n equaton (4). T retr and T succ represent the average delay for retransmsson and successfully transmsson. Detals for the dervaton of T retr and T succ are gven n [28]. Delay _ (1 P ) P tr retr T retr P succ T succ (4) Next, MBE estmates the throughput of a flow usng equaton (5), whch analyses the packet transmsson probablty and delay. The parameter Payload s the total nformaton transmtted durng tme perod from T 0 to T 1, and Delay_ denotes the average delay for successfully transmtted ndvdual packets. B T 1 Payload dt T T 0 Delay _ T 1 0 (5) Fnally, MBE derves a formula predctng the achevable bandwdth when TCP and flows co-exst n networks, as shown n equaton (6). The parameter w s the bandwdth weght factor snce TCP and have dfferent bandwdth requrements. TCP B w B ( 1- w) B TCP (6) MBE has been modeled, mplemented, and tested through smulatons and real lfe testng. The results show that MBE performs very well n condtons wth varable packet sze, dynamc wreless lnk rates and dfferent channel nose [28]. B. VoIP -Admsson Control The prncple of the VoIP-Admsson Control s to lmt the number of VoIP calls based on the predcted VoIP network

9 capacty so that the QoS of exstng VoIP flows wll not be degraded, whle mantanng hgh network bandwdth resource utlzaton. The VoIP transport capacty s defned as the number of VoIP flows whch can be supported by a wreless network for a gven average QoS level. The admsson polcy of the proposed VoIP-Admsson Control scheme s as follows. When a new VoIP call ntends to be sent over an IEEE network already carryng N VoIP flows, f the throughput of the N+1 flows exceeds the predcted VoIP capacty, then the new VoIP call s rejected; otherwse, t s accepted. VoIP-Admsson Control s deployed at the applcaton layer of the VoIP system and utlzes two types of nformaton: 1) network avalable bandwdth as t s estmated by MBE; 2) VoIP flow nformaton, as t s known, at the codecs ncludng codec bt-rate, packetzaton nterval and protocol header sze. The VoIP codecs nformaton depends on the applcaton deployed, but can be precsely gathered. Consequently, an accurate estmaton of the VoIP capacty s crtcal for an effcent admsson control. There are two steps performed by the VoIP- Admsson Control to compute the VoIP capacty. In step one, the bandwdth requred by an ndvdual VoIP flow s computed as shown n equaton (7). The parameter btrate s the voce packet coded btrate, pktintvl represents the voce packet nterval and headersze s the packet header sze. IndvdualFlowBandwdth headersze btrate pktintvl (7) In step two, the VoIP capacty s computed usng equaton (8). The Avalable Bandwdth refers to the overall network bandwdth resources whch can be used for the VoIP flows and s estmated by MBE usng equaton (6). The Indvdual Flow Bandwdth s the bandwdth requred by each VoIP flow whch s gven by equaton (7), VoIPCapacty N 1 AvalableBandwdth IndvdualFlowBandwdth (8) TABLE I VOICE STANDARD SPECIFICATIONS Codec Samplng Btrate Inter-packet Interval Voce Packet Rate (KHz) (Kbps) (ms) Sze (Bytes) ITU-T G ITU-T G LBC For example, we consder delvery of VoIP flows whch use the ITU-T R.G.711 encodng standard [33]. Table I presents the specfcatons of the wdely used VoIP codec. For nstance, ITU-T G.711 codec uses a voce samplng rate of 8000 samples per second and adopts non-unform quantzaton wth 8 bts to represent each sample, resultng n a content bt-rate of 64Kbps. There

10 s a drect relatonshp between the nter-packet delvery nterval and voce packet sze. Hgher values of nter-packet delvery nterval lead to larger voce packet sze. G.711 selects 20ms and 160bytes as default values for the nter-packet nterval and voce packet sze, respectvely. Accordng to [34] and [35], large voce packet sze mght result n hgher VoIP capacty but lower tolerance to packet loss and jtter, n comparson wth the case when shorter voce packet szes are used. Most VoIP packets are delvered over RTP//IP, so 40 bytes packet headers are often added to the payload. For G.711 VoIP servces, when the nterpacket nterval equals 20ms, VoIP flow bt-rate s 64Kbps and header sze s 40bytes, f the avalable bandwdth s 3Mbps, the VoIP capacty s 31 flows computed accordng to equaton (7) and equaton (8). VoIP scheme was desgned to use the wdely deployed constant bt-rate VoIP codec such as ITU-T G.711, ITU-T G.729 [36], LBC [37], etc. Accordng to equaton (7), three parameters are needed as nput: bt-rate, packet nterval and packet header payload. In the case of LBC, the bt-rate s constant for a certan nterval (20ms or 30ms). Take 20ms nterval for nstance, the bt-rate and packet header payload are 15.2Kbps and 38bytes, respectvely. Therefore, VoIP can be adapted to LBC codec straghtforwardly. C. VoIP- Farness The proposed MAC-layer VoIP-Farness scheme provdes QoS-based far channel access between downlnk and uplnk VoIP traffc. It takes the QoS parameters (.e. throughput, delay and loss) as nput and adapts the contenton wndow (CW) sze at the AP usng a stereotypes-based adaptve soluton. Detals of the prncple of the stereotypes-based adaptaton are presented next. 1) Stereotypes-based Data Structure The stereotypes-based structure s desgned to store nformaton about streams and provde ths data to the CW adaptaton scheme. Each stereotype (Th) s defned for a subgroup of streams and conssts of two components: a group of features F= (F 1, F 2,, F,, F m ) descrbng the stereotype and a group of suggestons S= (S 1,S 2,,S j,,s n ) that represent actons to be performed. Each feature F s assocated to a lst of lngustc terms LF = (LF 1, LF 2,,LF p,,lf q ). Each lngustc term LF p has a numerc value PF p between 0 and 1, representng the probablty that the feature F equals the lngustc term LF p for ths stereotype (Th). A smlar structure s defned for each suggeston S j, whch has also assocated the lngustc terms LS j = (LS j1, LS j2,, LS jr,, LF js ) and probablstc values PS jr. Table II and Table III present the group of features and suggestons for a stereotype. The Posson dstrbuton represents the probablty of a number of events occurrng durng a tme perod and s used to determne the probablty assocated wth the lngustc terms. The occurrences of events are ndependent from one-another. Equaton (9) shows the Posson dstrbuton functon where u s the shape parameter (from 0 to 15) and ndcates the mean and the varance of the dstrbuton durng a tme nterval. The nteger value x (x = 0, 1, 2,, n) represents a partcular event. By analyzng the shape of the Posson functon, a near normal dstrbuton s obtaned for u=7 across the (0, 15) nterval. The

11 selected value of u has been used by [38] for network parameters modelng, and resulted n very good results. The maxmum value of the normal dstrbuton s close to 0.15 (x=7, u=7) and the mnmum value s close to 0 (x=0 or x=15, u=7). Consequently, the nterval (0, 15) s consdered for the computaton of the Posson functon for all the stereotypes. Fg. 3 shows the example when VoIP uses fve stereotypes: Hgh Prorty (HP), Medum-Hgh Prorty (MHP), Medum Prorty (MP), Medum-Low Prorty (MLP), and Low Prorty (LP). It s notced that each stereotype assocates one Posson dstrbuton wth a mean value u k whch s obtaned by dvdng the nterval (0, 15) n fve equal segments and consderng ther mddle value. As shown n the Fg. 4, the peak value of the Posson functon ncreases when u k gets closer to zero pos (x, 1.5) pos(x, 4.5) pos (x, 7) pos (15-x, 4.5) pos (15-x, 1.5) Posson(X) Features X Fg. 4 Posson dstrbuton for fve stereotypes TABLE II GROUP OF FEATURES FOR A STEREOTYPE (Lngustc Term, Probablty) F 1 (LF 11, PF 11 ), (LF 12, PF 12 ),, (LF 1q, PF 1q ) F 2 (LF 21, PF 21 ), (LF 22, PF 22 ),, (LF 2q, PF 2q ) F m (LF m1, PF m1 ), (LF m2, PF m2 ),, (LF mq, PF mq ) Suggestons TABLE III GROUP OF SUGGESTIONS FOR A STEREOTYPE (Lngustc Term, Probablty) S 1 (LS 11, PS 11 ), (LS 12, PS 12 ),, (LS 1q, PS 1q ) S 2 (LS 21, PS 21 ), (LS 22, PS 22 ),, (LS 2q, PS 2q ) S m (LS m1,ps m1 ), (LS m2, PS m2 ),, (LS mq, PS mq ) pos( x, u) u x exp( u) x! (9) Consderng that feature F has a lst of lngustc terms, where the lst length s q, the probablstc values for each term PF j are computed as n equaton (10). The value s the ndex of the feature and j s the ndex of the lngustc term.

12 x j PFj Average ( pos( x, u)) 15 / q( j 1), 15 q / j j (10) The stereotype-based structure s ntalzed and updated usng the followng three stage process: Stage 1: User Classfcaton The purpose of the stereotype-based classfcaton s to determne the stereotype class the stream belongs to and wth what probablty. Equaton (11) presents the format of data assocated to a data stream D, where F s the name of the th feature and LF V represents the lngustc value of the th feature. D (( F, LFV ),( F 1 2, LFV 2),...,( F m, LFV 1 m )) (11) A degree of match between the stream and each stereotype class s computed as n equaton (12) based on the probablty theory. M ( Th) p( Th F p( Th F 1 1 LFV,..., F LFV )... p( Th F 1 1 m LFV m m ) LFV The computaton of each factor s computed usng the Bayes rule, as shown n equaton (13). It s consdered that all the stereotypes have the same probablty dstrbuton, therefore, p(th) s the recprocal of the number of stereotype classes. m ) (12) p( Th F LFV ) p( F LFV Th) p( Th) p( F LFV ) PFV p( Th) p( F LFV ) (13) Stage 2: Suggeston Determnaton After the stream s assocated wth stereotype classes, the suggeston determnaton s performed to determne the best CW sze to be allocated to the AP. Frst, for each stereotype, the strength of each suggeston has to be computed by consderng the probablty for the stream D to belong to ths stereotype class, as n equatons (14), where S s the name of the th suggeston and LSV represents the lngustc term of the th suggeston. p( S LSV Th) p( S LSV Th) M( Th) (14) Second, the combnaton of the suggestons of each stereotype class s computed by equaton (15), based on the probablstc theory. P(E n ) s the probablty of an event E n to appear and p (E 1 &E 2 & &E n ) represents the probablty for events E 1 &E 2 & &E n to appear smultaneously.

13 p( E1 & E2 &... En ) p( En ) [1 p( En )] P( E1 & E2 &... En 1) (15) Stage 3: Update The stereotypes are updated usng a four-step algorthm: Track the user related parameters regardng each feature (F ) of the stereotype. For each stereotype (Th), re-calculate the probablty (PF p ) value that s assocated wth the lngustc term LF p. Repeat the User Classfcaton process descrbed before, n order to determne the stereotype class the current stream belongs to and wth what probablty. Repeat the Suggeston Determnaton process to produce weghted suggestons based on the probablty wth whch the stream belongs to each stereotype classes. Merge the suggestons found wth the prevously determned suggestons to determne the adaptaton approach. 2) CW Adaptaton usng the Stereotype-based Structure Fve stereotype classes are used to represent fve farness levels between downlnk and uplnk: Bad, Poor, Normal, Good, and Excellent. Theoretcally, there can be any number of farness levels. In ths paper, fve farness levels are selected n order to correspond to the fve Mean Opnon Score levels n the ITU-T Recommendatons P.800 [39]. Three QoS performance parameters, Throughput down/up, Delay down/up, and LossRate down/up are modeled as stereotype features, representng throughput rato, delay rato and loss rato between the downlnk and uplnk communcaton channels, respectvely. The computaton of the downlnk/uplnk rato for each QoS parameters s descrbed n the next secton. The lngustc terms for the features of each stereotype are denoted usng fve ranges representng the possble ratos between the downlnk and uplnk as follows: >1.5 (LF 1 ), (LF 2 ), (LF 3 ), (LF 4 ), <0.6 (LF 5 ). Parameter ndcates the th stereotype. The ntal CW sze for the AP and each wreless staton s selected randomly from 0 to CW mn (equal to15) accordng to standards. The dea of adaptng the AP s CW usng the stereotype-based structure s to assocate a CW sze accordng to stereotypes suggestons. Addtonally, the CW range (.e., , as specfed n standards) s dvded nto four levels representng fve suggeston lngustc terms as follows: 0-15 (LS j1 ), (LS j2 ), (LS j3 ), (LS j4 ), (LS j5 ). LS j1 represents the ntal CW range. LS j2, LS j3, LS j4, and LS j5 are the equally dvded CW ranges, each of whch s set to 251. Parameter j ndcates the j th stereotype. The adapted CW sze of AP s then computed usng the three stage process ntroduced n the prevous secton: User Classfcaton, Suggeston Determnaton and Update. The adapted CW sze of AP s then computed usng the three stage process ntroduced n the prevous secton: User Classfcaton, Suggeston Determnaton and Update. VoIP dvdes the orgnal CW range (.e ) nto fve levels (.e. 0-15, , , , ). that are assocated to the fve stereotypes levels ( Bad, Poor, Normal, Good, and Excellent ), respectvely.

14 The fve stereotypes levels are mapped to fve downlnk/uplnk QoS-parameters (delay, throughput, delay) rato: >1.5, , , , <0.6. The nstant CW sze of AP s determned by selectng a random the md value from the suggesteda CW nterval. For nstance, f the QoS-parameters rato acheved by downlnk/uplnk flows falls nto , the related CW nterval s and the md value ) Computaton of Stereotype Feature Values The proposed CW adaptaton scheme collects transmsson-related nformaton at AP as well as at the wreless statons. A feedback mechansm s employed to carry the nformaton collected at the wreless statons to the AP. The feedback nformaton s delvered usng RTP Control Protocol (RTCP) protocol [40], whch allows for defnng addtonal user defned packet types n the Recever Report packet header. Ths nformaton conssts of throughput downlnk/uplnk rato, delay downlnk/uplnk rato, and packet loss rate downlnk/uplnk rato. Throughput Downlnk/uplnk Rato The throughput downlnk/uplnk rato s consdered far when the throughput n both drectons has equal values and therefore Throughput down/up rato equals one. Equaton (16) llustrates how the downlnk/uplnk throughput rato s computed. Throughput down/ up N 1 Throughput Throughput AP STA (16) Equaton (16) makes use of the throughput at the AP (Throughput AP ) and the aggregaton of the throughput at the wreless statons (Throughput STA ), where ndcates the th wreless staton and N s the number of wreless statons. Both Throughput AP and the overall throughput at the wreless statons can be measured at the MAC layer of the AP. Delay Downlnk/Uplnk Rato The downlnk and uplnk delay dstrbuton s far when the two communcaton drectons wll process the same amount of traffc durng a sample nterval. For example f the packet sze s dentcal, t s far to have the AP sendng N packets to the statons (downlnk) and have the N wreless statons sendng N packets to the AP (uplnk) n the same tme perod. In order to acheve delay downlnk/uplnk farness, t s noted that there are three types of delay for packet transmssons va wreless: 1) Queung delay; 2) MAC delay; 3) Propagaton delay. Queung delay represents the total duraton of tme that packets have to wat n the queues. MAC delay s caused by the contenton mechansm of CSMA/CA protocols, whch may also nclude some uplnk-downlnk unfarness. However queung delay s by far the largest delay that causes delay unfarness between downlnk and uplnk [19], [41] and therefore MAC delay s not consdered n ths paper. Propagaton delay s dependent on the dstance and sgnal propagaton speed only. In the VoIP system, the wreless propagaton delay s the same between downlnk and

15 uplnk because they use the same medum and as the packet sze s the same, t does not nfluence the downlnk/uplnk farness. The queung delay s determned by many factors such as queue arrvng rate, queue servce rate, current queue sze, etc. Snce there s a desre that the proposed algorthm be deployed at the AP wthout modfyng the wreless statons, the current queue sze of the th wreless staton QSze STA s estmated usng equaton (17). The parameter AvgPktSze s the average packet sze receved from the th wreless staton durng the sampled nterval. λ STA represents the arrval rate of the packets enterng the queue, and depends on the VoIP encodng scheme. For nstance, a 64kbps VoIP traffc mples λ STA =64kbps. The VoIP encoded nformaton s delvered to AP va the feedback mechansm. NumRcvdPkts s the number of packets receved by the AP from the th wreless staton n the sample tme perod. Tme s the samplng tme duraton selected to nvestgate the queue state. QSze STA AVGPktSze ( Tme STA NumRcvdPkt s ) (17) The burst arrval of packets results n a random queue sze. If the sample nterval s too small, t s possble that t may be no packet transmssons due to the bursty nature of traffc. Otherwse, too bg nterval value reduces the update frequency leadng to naccuracy n the queue sze estmaton. The samplng nterval s selected based on the Nyqust theorem [42], gven n equaton (18), where w s the sgnal frequency. The purpose of computng an optmal sample nterval s to allevate the alasng phenomenon due to the traffc busrtness. Tme SamplngInterval 1/(2 w) (18) Delay down/ up N 1 QDelay QDelay AP STA N 1 QSze QSze AP STA / AP / STA (19) The delay downlnk/uplnk rato s gven n equaton (19), where QDelay AP and QDelay STA are the average queung delay at the AP and th wreless staton, respectvely. QSze AP and QSze STA are the amount of packets (n bts) watng n the queue at the AP and th wreless staton, respectvely. µ AP and µ STA are the servce rate of the queue at the AP and th wreless staton, respectvely (.e. the rate at whch bts leave the queue). The values of QSze AP, µ AP and µ STA can be montored and measured at the AP. The values of QSze STA are computed usng equaton (17) and equaton (18). Packet Loss Rate Downlnk/uplnk Rato The end to end packet loss s caused by the followng factors: 1) Queue Drop: Packets can be dropped at the queue dependng on the queung management algorthms adopted. For nstance, n the Frst Come Frst Servce (FCFS) queues such as DropTal [43], packets are dropped when the queue has flled ts capacty; n Random Early Detecton (RED) queues [44], packets are dropped wth certan probablty dependng on the queue sze; 2) Channel Error: packets can be lost due to the wreless channel error; 3) Retransmsson Lmt: when packet retransmsson reaches a retry lmt defned by the MAC, the packet s

16 dropped. 4) Collson: collsons occur when multple wreless statons (uplnk) attempt to transmt the packets smultaneously; packets affected by collsons are dropped. There are no collsons among the downlnk flows snce the AP s the unque DCF object generatng traffc n the downlnk mode. The packet loss rate downlnk/uplnk rato s gven n equaton (20) and (21). BERLoss AP BER (20) AP LossRate down/ up QDropRate AP BERLoss N 1 Loss AP STA RETRANLoss AP (21) The parameters QDropRate AP, BERLoss AP, and RETRANLoss AP represent the packet loss at the AP caused by queue drop, channel error and retransmsson lmt, respectvely. QDropRate AP and RETRANLoss AP are captured at the MAC layer of the AP, and BERLoss AP s computed usng equaton (20), where µ AP s the servce rate of the AP queue. The parameter Loss STA s the packet loss rate of the th wreless staton whch s measured at the AP based on packet sequence numbers. IV. EXPERIMENTAL TESTING SETUP The performance of the proposed VoIP-Admsson Control and VoIP-Farness schemes were evaluated va both real-lfe and smulatons. Testng setup s descrbed next, ncludng the characterstcs of the VoIP traffc used, test-bed confguraton, and expermental scenaros. A. Real Lfe Test-bed Setup Real lfe measurements were conducted usng SIPp [45], whch s an open-source SIP traffc generaton tool that supports generaton of multple VoIP calls from callers to callees. The callers and callees were deployed n HP Pavlon Entertanment laptops and assocated wth User Agent Clent (UAC) and User Agent Server (UAS), respectvely. UAC and UAS are basc SIPp user agent nstances that support establsh and release VoIP calls. UAC transmtted SIP messages usng RTP/ protocol. The voce codec used was ITU-T R.G.711 [33] wth a default nter-packet nterval of 20ms. Wreshark [46] software was used to measure the throughput at UAS. A LnksysWRV210 access pont wreless router was used to support IEEE b WLAN-based network. In these experments, PHY data rate was 11Mbps and all the statons were statonary. RTS/CTS mechansm was dsabled due to the short packet sze of the VoIP servce.

17 Server 1 Clent 1 Server 2 router AP Clent 2 Clent N Server N Fg. 5 Test bed topology Fg. 6 Random topology n smulaton B. Smulaton Test-bed Setup Smulaton-based tests were performed usng Network Smulator NS-2 3. The smulaton topology used the wred-cum-wreless dumbbell topology wth a 100Mbps bandwdth and 20ms delay wred bottleneck, as shown n Fg. 5. Each wreless staton receves a sngle VoIP flow from a wred staton through the bottleneck lnk and the same IEEE AP. Constant Bt Rate (CBR) VoIP traffc was generated usng the ITU-T R.G.711 codec [33], wth payloads of 160 bytes/packet. The bt-rate of CBR was set to 64kbps representng an nter packet nterval of 20ms. DropTal [41] queue wth a lmt of 100 packets was set to each wreless staton. The RTS/CTS mechansm was dsabled as voce packets are small. The MAC layer parameters were confgured accordng to the b specfcatons [47], where CW mn =31, CW max =1023, DCF Interframe Space (DIFS)=50µs, Short Interframe Space (SIFS)=10µs, and slot tme=20µs. The ntal CW sze of AP was selected randomly from 0 to 31. Two 3 Network Smulator NS-2. [Onlne]. Avalable:

18 addtonal wreless patches are deployed n the NS-2: NOAH 4 and Marco Fore patch 5. NOAH (No Ad-Hoc) was used for smulatng the nfrastructure WLAN envronment, whereas Marco Fore s patch provdes a more realstc wreless network envronment. As shown n Fg. 6, the scenaro consders multple moble nodes located at random locatons around AP. Darker colors area ndcates areas wth hgher bt-rates and stuated closer to the AP. Accordng to the IEEE b specfcaton, the data rate degrades step-wse takng values at at four levels: 11Mbps, 5.5Mbps, 2Mbps, and 1Mbps. The dstance between moble nodes and AP ranges from 30m to 120m n order to test the effect of varable channel qualty. C. Expermental Scenaros In order to study the performance of both proposed VoIP-Admsson Control and VoIP-Farness solutons, three separate scenaros were desgned. 1) Scenaro 1. Ths scenaro ams to evaluate the accuracy of the VoIP capacty provded by the VoIP-Admsson Control. VoIP was deployed n the smulaton test bed. The real lfe test-bed was used to provde measured capacty. The estmated VoIP capacty was computed usng equaton (5) and then compared wth the measurement results. The SIPp UAC was confgured to generate VoIP flows at a rate of 0.2 cps (calls per second). The number of VoIP flows was ncreased up to 50. ITU-T R.G.711 voce codec was used wth the packetzaton nterval ncreased from 10ms to 50ms wth step of 10ms. Two performance metrcs were studed: 1) the estmated and measured VoIP capactes wth varable nter-packet ntervals; 2) one-way delays of VoIP flows when VoIP and the orgnal IEEE protocol wth no admsson control mechansm were employed n turn. Notably, the VoIP capacty from the real lfe measurement s obtaned accordng to ITU-T R.G.114 [48], whch recommends a maxmum of a 150ms one-way delay for VoIP. Therefore, assumng the VoIP capacty from the measurement equals M, whch ndcates that delays experenced by all the M VoIP flows are below 150ms. 2) Scenaro 2. Ths scenaro nvestgates the performance of the VoIP-Farness module. VoIP was modeled and compared aganst the orgnal protocol and a state-of-the-art farness scheme, Dynamc-CW [17]. The number of wreless statons ncreased from 0 up to the VoIP capacty estmated n scenaro 1. VoIP performance was studed usng Jan s farness ndex [49] n terms of delay, throughput, and loss. 3) Scenaro 3. The thrd scenaro studes the nfluence of VoIP admsson control mechansm and farness scheme on VoIP qualty n terms of the E-model [50] whch descrbes VoIP qualty. An experment was conducted based on test scenaro 1, where each wreless staton mantans a sngle VoIP flow. The number of wreless statons N n ths study s ncreased up to NOAH NS-2 extenson, 5 M. Fore patch,

19 V. RESULT ANALYSIS The three expermental scenaros n secton IV were performed separately to study the performance of VoIP and ts major two contrbutons: the admsson control scheme and the farness algorthm. Result analyze of each experment are presented next n detals. The real-lfe test bed was bult to measure the actual VoIP capacty n order to evaluate the performance of VoIP admsson control module. In Fgure 7 and Table IV, the measured VoIP capactes are obtaned from real-lfe mplementaton and tests and whle the estmated VoIP capactes are taken from smulatons. A. VoIP- Admsson Control Ths secton studes the performance of VoIP capacty estmaton and admsson control algorthm, accordng to scenaro 1. Fg. 7 and Table IV present VoIP capacty wth varable nter-packet nterval for ITU-T R.G.711 and shows that there s a good match between analytcal results (VoIP) and measurement results. Specfcally, the capacty predcted by VoIP s optmstc n comparson wth the real lfe measurement. For nstance, n the case of codec nterval equals 20ms, the VoIP capacty acheves 24 and 22 for VoIP and measurement, respectvely. Ths can be explaned by two reasons: 1) the 150ms delay constrants n real lfe measurements are not ncluded n VoIP (as ndcated n scenaro 1); 2) varous real envronmental factors (e.g. fadng, shadowng, etc) mpact the measurements. Fg. 7 also shows that larger packetzaton nterval results n hgher VoIP capacty due to hgher channel effcency (e.g. lower overhead and less nterference). For nstance, when the codec nterval equals 40ms, the VoIP capacty ncreases by 62.5% and 63.6% for VoIP and measurement, respectvely, n comparson wth that of 20ms. TABLE IV VOIP CAPACITY WITH VARIABLE INTER-PACKET INTERVAL Inter-packet VoIP Capacty Interval (ms) VoIP Measurement VoIP Capacty Real lfe measurement VoIP Interval (ms) Fg. 7 VoIP capacty wth varable nter-packet nterval for ITU-T R.G.711

20 TABLE V ONE WAY DEAY OF VOIP SERVICE WITH INCREASING NUMBER OF VOIP FLOWS (CODEC: ITU-T R.G.711, CODEC INTERVAL: 20MS) N IEEE wth VoIP (ms) IEEE wthout Admsson Control (ms) IEEE wthout Admsson Control IEEE wth VoIP delay (ms) Number of wreless statons (N) Fg. 8 One-way delay of VoIP servce wth ncreasng number of VoIP flows (codec: ITU-T R.G.711, codec nterval: 20ms) Fg. 8 and Table V show the average one-way delay experenced by an ndvdual VoIP flow usng IEEE wth VoIP admsson control and classc IEEE wthout admsson control. The test s also based on scenaro 1 except that the maxmum number of VoIP flows was reduced. As ndcated n Fg. 7, the VoIP capacty s below 30 when the nter-packet nterval s 20ms, therefore, the number of VoIP flows ncreased up to 30. In general, the delay ncreases as the number of VoIP flows (N) ncreases for both VoIP admsson control and IEEE wthout admsson control. In the case of VoIP, the oneway delay remans below 150ms whch s very good. Ths s as all the ncomng flows are rejected when N exceeds 24, whch s the VoIP capacty as shown n Fg. 7. In the case of orgnal IEEE wthout admsson control, the one-way delay ncreases dramatcally when N=22, whch s the VoIP capacty, as ndcated n Fg. 7.

21 TABLE VI JAIN S FAIRNESS INDEX ACHIEVED BY , DYNAMIC-CW, AND IVOIP IN TERMS OF DELAY, THROUGHPUT, AND LOSS Delay Throughput Loss N Dynamc- IEEE Dynamc- IEEE Dynamc- IEEE VoIP VoIP VoIP CW CW CW B. VoIP-Farness Ths secton presents the expermental results based on scenaro 2. Specfcally, as shown n scenaro 1, the VoIP capacty predcted by VoIP-Admsson Control module s 24 when codec nterval equals 20ms. Therefore, the number of VoIP flows s ncreased up to 24 n scenaro 2. In order to evaluate the performance of VoIP-Farness module, downlnk and uplnk farness was measured usng the Jan s farness ndex [49] for all QoS parameters consdered. Let Q D (=1, 2,, M) and j Q U (j =1, 2,, N) represent the QoS parameters (throughput, delay, packet loss rate) of the the th downlnk flow and j th uplnk flow. The Jan s farness ndex n terms of QoS of the downlnk and uplnk traffc s gven n equaton (22), where separate parameters FI down/up are computed for throughput, delay, and packet loss rate, respectvely. FI down/ up ( ( M N)( M N j 2 QD QU ) 1 j1 M N 2 ( QD ) 1 j1 ( Q j U ) 2 ) (22) Fg. 9, Fg. 10 and Fg. 11 present the Jan s farness ndex n terms of QoS parameters, delay, throughput and loss, respectvely. Table VI shows the expermental results regardng wth the three fgures. The results from VoIP are compared wth that of the orgnal IEEE protocol and Dynamc-CW. The expermental results show that the farness level for the three schemes decreases along wth ncreasng number of wreless statons. Moreover, the followng conclusons are made: 1) the average farness levels acheved by VoIP are 25.8%, 16%, and 18.9% hgher than , n terms of delay, throughput, and loss, respectvely; 2) the average farness levels acheved by VoIP are 21.5%, 8.5%, and 10.5% hgher than Dynamc-CW, n

22 terms of delay, throughput, and loss, respectvely. It s clear that, the most sgnfcantly mproved QoS parameter s delay, whch s crtcal for VoIP servces. C. VoIP Qualty Ths secton studes the effect of the proposed VoIP admsson control mechansm and farness scheme on VoIP qualty. Scenaro 3 was used for the test. ITU-T E-Model [50] s developed to evaluate the voce applcatons n heterogeneous crcut/packet-swtched networks. A basc result of the E-Model s R-factor, whch evaluates voce qualty by takng nto account both physcal equpment mparments and perceptual effects to the equpment mparment. R-factor ranges from the worst case of 0 to the best case of 100 and has been related to Mean Opnon Score (MOS). The computaton of the R-factor s smplfed accordng to [51] and s gven n equaton (23). I d represents the mparment caused by mouth-to-ear delay ncludng codec delay, network delay and playout delay. I ef s assocated wth losses due to codecs and network. Note that, equaton (23) does not mply that I d and I ef are unrelated only that ther mpacts on the mparments are separable. R I d I ef (23) Snce I d and I ef are dffcult to obtan n real tme, R-factor s smplfed as functons of delay and loss measurements only [51] for the case of G.711 codec, as shown n equaton (24). R d 0.11 ( d 177.3) H ( d 177.3) 30 ln(1 15 e) (24) 1.0 Jan's Farness Index (Delay) IEEE Dynamc-CW VoIP Number of wreless statons (N) Fg. 9 Jan s farness ndex n terms of delay

23 Jan's Farness Index (Throughput) IEEE Dynamc-CW VoIP Number of wreless statons (N) Fg. 10 Jan s farness ndex n terms of throughput Jan's Farness Index (Loss) R-factor IEEE Dynamc-CW VoIP Number of wreless statons (N) Fg. 11 Jan s farness ndex n terms of packet loss rate downlnk(ieee ) uplnk(ieee ) downlnk(dynamc-cw) uplnk(dynamc-cw) downlnk(voip) uplnk(voip) Numer of wreless statons (N) Fg. 12 R-factor values of downlnk and uplnk flows n , Dynamc-CW and VoIP

24 Parameters d and e refer to the one-way delay (n mllseconds) and the total loss probablty (e ranges from 0 to 1), respectvely. H(x) s an ndcator functon: H(x) = 0 when x<0 and H(x)=1 when x 0. Equaton (25) and equaton (26) [51] show the formulas for the computaton of both d and e. d d codec d playout d network (25) e e (1 e ) e network network playout (26) TABLE VII R-FACTOR AND MOS VALUES OF DOWNLINK AND UPLINK FLOWS IN , DYNAMIC-CW, AND VOIP VoIP Dynamc-CW IEEE N downlnk uplnk downlnk uplnk downlnk uplnk R MOS R MOS R MOS R MOS R MOS R MOS d codec s the codec delay whch equals 20ms for ITU-R.G.711 codec. d playout s the playout delay caused at decoder s buffer and s set to 60ms by default. e playout s the loss probablty due to overflow or underflow at decoder s buffer and s set to by default. Parameters d network and e network represent transport level delay and loss probablty, respectvely, whch are measured n real tme. Fg. 12 and Table VII show that VoIP outperforms both IEEE and Dynamc-CW n terms of R-factor and mean opnon score (MOS) values between downlnk and uplnk flows. The mappng between R-factor and MOS s gven by equaton (27) [50]. 6 MOS R R( R 60) (100 R) 7.10 (27) When the number of wreless statons (N) exceeds the normal VoIP capacty (.e. N=22), R-factor values acheved by downlnk and uplnk flows for both and Dynamc-CW decrease dramatcally. For nstance, when N=30, R-factor values of downlnk and uplnk for s reduced by 40.5% and 35.8%, respectvely, n comparson wth the case when N=22. Whereas for VoIP, there s no sgnfcant decrease n R-factor due to the admsson control mechansm adopted. For nstance, the number of VoIP