A SIMULATION ANALYSIS OF AGGREGATION STRATEGIES IN A WF 2 Q+ SCHEDULERS NETWORK

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1 A SIMULATION ANALYSIS OF AGGREGATION STRATEGIES IN A WF 2 Q+ SCHEDULERS NETWORK R. G. Garroppo, S. Gordano, S. Nccoln, F. Russo {r.garroppo, s.gordano, s.nccoln, f.russo}@et.unp.t Department of Informaton Engneerng Unversty of Psa Va Dotsalv Psa - Italy Tel , Fax Abstract -- The paper presents an analyss n a DffServ network scenaro of the achevable QoS (Qualty of Servce) performance when dfferent aggregaton strateges between vdeo and voce traffc flows are consdered. Each network node of the analyzed DffServ scenaro s represented by a Worst-Case Far Weghted Far Queueng scheduler wth a shaper obtaned usng a Shaped Startng Potental Far Queueng. The system of each node, referred n lterature as WF 2 Q+ scheduler, permts to guarantee the solaton among the dfferent PHB (Per Hop Behavor) servce classes, mantanng the multplexng gan. The parameters settng of the WF 2 Q+ scheduler s also dscussed. In partcular, the necessary network resources, estmated by the WF 2 Q+ parameters settng obtaned consderng the aggregaton of traffc sources belongng to the same servce class, are compared wth those estmated on a per flow bass. The hgher gan achevable usng the frst approach wth respect to the second one, s also qualtatvely hghlghted. The smulaton results, presented n the paper, evdence the possble problems that can be rased when voce traffc s merged wth vdeo servce traffc. As a consequence, the paper results suggest to consder n dfferent servce class queues the two knds of traffc. Keywords -- WF 2 Q+ scheduler, LBAP traffc characterzaton, DffServ archtecture, voce model, QoS parameters 1. Introducton A key challenge of the current telecommuncaton age s represented by the developng of new archtecture models for IP networks n order to satsfy the recent QoS requrements of nnovatve IP-based servces (e.g. IP Telephony and vdeoconferencng). At present, the ISP (Internet Servce Provder) often provde the same servce level ndependently from the traffc generated by ther clents. Takng nto account the transformaton of Internet to a commercal nfrastructure, t s possble to understand the need to provde dfferentated servces to users wth wdely dfferent servce requrements. In ths framework, the DffServ approach s the most promsng for mplementng scalable servce dfferentaton n IP networks. The scalablty s acheved by consderng the aggregate traffc flows and condtonng the ngong traffc at the edge of the network. Aggregaton obvously decreases the complexty of traffc control n the core network, but t produces some unwelcome effects, such as lock-out or full-queues phenomena, whch contrbute to ncrease end-to-end delay and jtter of the traffc flow of a sngle servce. These two phenomena take place respectvely when few flows monopolze queue space preventng other connectons from gettng n the queue and when t s not possble to mantan the queues non-full. Hence, the effects of the aggregaton mechansms on the QoS parameters of the dfferent aggregated flows need to be further analyzed. The frst works n ths feld have hghlghted relevant concepts to support traffc aggregaton [1], however a stll open ssue s what knd of aggregaton strateges s better to carry out. To ths am we nvestgate traffc aggregaton strateges because there s stll no clear poston on what s the better confguraton (standardzaton organsms say anythng regardng ths matter). On the other hand recent publcaton [2] suggests to dvde network traffc n only two servce classes (e.g. real-tme and non real-tme) but t seems, from our pont of vew, a lttle bt restrctve wth respect to dfferent traffc features. In the paper, we analyze the mpact of the aggregaton of real-tme vdeo and voce traffc n the same servce class (hence, n the same queue n a per-servce queueng system) on the expermented QoS parameters of the dfferent flows. The QoS concept used n ths work s to be dentfed wth the whole set of propertes whch characterze network traffc (e.g. n terms of resource avalablty, end-to-end delay, delay jtter, throughput and loss probablty). The results obtaned n ths scenaro are then compared wth those obtaned consderng real tme vdeo and voce as separate traffc flows. The DffServ archtecture [3] s a good startng pont but t s useless f there s no teletraffc engneerng background able to provde the needed dfferentaton. Therefore we should take nto account also schedulng dscplnes and ther dmensonng, n order to understand f they may affect results (changng schedulng dscplne change the way the flows are treated). Hence, we have frstly chosen to use one of the best work-conservng

2 schedulng algorthms (WF2Q+) nstead of a non workconservng one used n other analyss [4]. The choce derves from the assumpton that the best the schedulng algorthm s (keepng acceptable ts complexty) the better treatment a flow receves n terms of low end-to-end delay and jtter. Moreover, the parameters settng of the consdered schedulng dscplne has been analyzed as descrbed s n Secton 4. In the analyss, we use as traffc characterzaton approach, the LBAP (Lnear Bounded Arrval Processes) theory [5]. Furthermore, we nvestgate the effects on LBAP characterzaton of the multplexng of the vdeo traffc, nstead of takng nto account the smple sum of the traffc descrptors obtaned wth the sngle source. By means of smulaton analyss we evaluate f the multplexng gan derved from characterzaton of aggregated traffc does not affect the QoS parameters. The rest of the paper s organzed as follows. In Secton 2 we present the smulaton scenaro whle n Secton 3 we descrbe the voce source model, the vdeo and data traffc taken nto account n the smulaton analyss. In Secton 5, the results are dscussed whle Secton 6 summarzes the man results presented n the paper. 2. Smulaton Scenaro Our smulaton scenaro manly reflects the topology of a DffServ doman of an IP network. The smulaton scenaro s mplemented usng the OPNET Modeler vers. 6.0.L, a powerful CAMAD (Computer Aded Modelng and Desgn) tool used n modelng communcaton systems and n analyzng network performance. The consdered scenaro s shown n Fg Fg 2.1: Smulaton scenaro The network model s represented by edge and core routers, each one havng a work conservng scheduler that permt to realze the solaton of the enterng flows, based on performance guarantees. The schedulng dscplne s a Worst-Case Far Weghted Far Queung wth the addton of a Shaper, obtaned usng a Shaped Startng Potental Far Queung (SSPFQ), denoted as WF 2 Q+ [6]. The WF 2 Q+ s a GPS (Generalzed Processor Sharng) approxmatng servce dscplne wth hgh farness propertes and relatvely low mplementaton complexty. Moreover, n order to smulate a sngle Dffserv doman, we mplement at the edge router the classfer and the marker necessary to assocate each packet to the selected PHB. Based on ths classfcaton and markng, each packet receve the sutable forwardng treatment by the core routers. The traffc sources taken nto account n the smulaton scenaro, are the most heterogeneous possble because we want to analyze the performance of a real network; t must ntegrate the carryng of vdeo, voce and data traffc. Hence, descrbng the scenaro shown n Fg. 2.1 n more detals, every block named as host 1, 4 and 7 contans 15 voce sources, whle every block named as host 3, 6 and 9 contans a vdeo source. The remanng hosts contan data module, whch smulate best-effort traffc. The statstcs we have collected concern the most sgnfcant QoS parameters of real-tme servces,.e. endto-end delay and jtter delay, whch are evaluated consderng the connecton among the dfferent sources and the destnaton node shown n Fg Source models In the smulatons, we adopt a model only for the voce sources, whle for the other knds of traffc we consder actual traffc data. The model used for the voce sources conssts n an On-Off model, suggested by the typcal behavor of a voce source wth VAD (Voce Actvty Detecton): t s actve or nactve dependng on the talker s speakng or slent. Assumng that no compresson s appled to voce sgnal, durng actve perods the source transmts at the constant bt rate of ν=64 Kbps (ths corresponds to a standard PCM codec wth VAD). In-depth analyses of ths traffc source, shown n lterature, have emphaszed that the dstrbuton of actve and nactve perods lengths can be approxmated by an exponental functon [7], wth mean values respectvely equal to T on =350 msec and T off =650 msec. The packet sze s 64 bytes, and consderng the bt rate and the header overhead (40 bytes takng nto account the RTP/UDP/IP header) the source generates one packet every 3 msec. Vdeo flow Mean_rate Peak_rate GOLDFINGER ASTERIX SIMPSONS Table 3.1 Statstcal parameters of consdered vdeo sources The traffc data used for the vdeo sources are descrbed n [8], where also ther statstcal analyss s presented. They have been obtaned collectng the output of an MPEG-1 encoder loaded by dfferent sequences of moves half an hour long. Some relevant statstcal parameters of the traffc data used, named Goldfnger,

3 Asterx and Smpsons, are summarzed n Table 3.1. The vdeo packets are produced at applcaton level, dvdng the number of bytes produced by the encoder n the frame perod, T=1/24 sec, n consecutve packets of sze equal to 1500 bytes (n ths case an MTU, Maxmum Transfer Unt, of 1500 s supposed). Moreover, n the smulaton we consder the 40 byte of overhead, related to the UDP/RTP/IP header, assumng that every packet transports 1460 byte of the traffc data regstered at the output of the encoder. Fg An example of packet fragmentaton at applcaton level The tme nterval between the generaton of the consecutve packet n each frame perod, has been consdered determnstc and equal to T/N, where N s the number of packets needed to transport all the bytes produced by the encoder durng a frame perod. As an example, Fg. 3.1 presents a case where 3200 bytes are necessary for the encodng of a frame; at applcaton level we dvde the frame n three packets, whch are sent wth tme nterval equal to T/3 sec. The Best Effort sources have been obtaned consderng the traffc data acqured at the Faculty of Engneerng of the Unversty of Psa. In partcular, we consder the traffc exchanges by the Faculty of Engneerng wth the external world (essentally other Unversty stes and Internet) by means of an ATM network at 155 Mbps [9]. The peak rate of the consdered traffc s equal to 11 Mbps, whle a mean rate of only 400 Kbps has been observed; the hgh peak-to-mean rato s a clear evdence of the hgh burstness of the data traffc. Durng the data acquston both the arrval tme and the sze of each packet have been regstered. Hence, n ths case the packet generaton process to use n the smulatons s drectly obtaned from the traffc data. 4. Parameters Settng In order to set the scheduler parameters, we frst characterze the traffc sources by mean of the LBAP approach. The traffc characterzaton of the sngle source s obtaned by the parameters (b, ρ) where b s ndcated as bucket sze and ρ as token rate. The physcal nterpretaton of the LBAP parameters can be understood, consderng that the number of bytes produced by a sngle source n a tme nterval of (0,τ), A(τ), s upper bounded by A ( τ ) b + ρτ τ>0 The results presented n [10] permts to have an upper bound for the end-to-end delay expermented by the traffc when t traverses through a Latency Rate scheduler network, as that consdered n our smulaton scenaro (.e. WF 2 Q+); estmaton of WF 2 Q+ latency term s descrbed n [11]. In partcular, the delay ntroduced by a sngle node to a packet belongng to the -th flow, characterzed by the LBAP parameters (b, ρ ) s upper bounded by (n the followng we suppose that for the -th flow a servce rate equal to ρ s allocated) b D + Θ ρ where Θ represents the latency of the scheduler, defned as Θ = L,max ρ L + C max. L,max and L max respectvely represent the maxmum packet sze of the -th flow and of the global traffc arrvng to the scheduler, whle ρ and C, are the servce rate allocated to the -th flow and the global output servce rate respectvely. Extendng the analyss to a network of K WF 2 Q+ schedulers, the end-to-end delay expermented by a sngle packet belongng to the -th flow s upper bounded by b K j D + j = 1Θ, ρ j where Θ ndcates the latency of the j-th node evaluated for the -th flow. The end-to-end delay bound has been obtaned consderng the worst-case analyss, whch s more conservatve wth respect to expermented end-to-end delay. Furthermore, as shown n [12], also the LBAP traffc characterzaton s conservatve wth respect to the statstcal modelng approaches. These two consderatons leads us to assume that the maxmum end-to-end delay of the -th flow can be upper bounded smply by D. ρ Ths hypothess permts to establsh the buffer sze and the guaranteed rate to set n the scheduler, smply evaluatng the LBAP curve of the -th flow. The smulaton results that wll be presented n Secton 5 pont out the goodness of our assumpton, showng the very conservatve nature of the worst-case analyss. Consderng the above hypothess, the procedure used to set the scheduler parameters conssts n evaluatng the LBAP curve and n fndng the pont where ths curve ntersect the straght lne b = ρd, where D represents the maxmum delay fxed for the consdered source. Fgure 4.1 shows the presented approach n the case of the vdeo sources. In partcular, n the fgure we can observe the LBAP curves for three dfferent vdeo sources, and the curve related to the traffc obtaned aggregatng these sources. Moreover, assumng a b

4 maxmum end-to-end delay of 200 msec, we can observe the relate straght lne and the ntersecton ponts wth each LBAP curves that, as descrbed above, gve the couple (ρ, b ) to consder n the settng of scheduler parameters. set the scheduler parameter assumng a utlzaton factor of the lnk equal to 0.9. In more detals, we evaluate the sum of ρ, ρ tot, and fx the rate of the output lnk equal to ρ tot /0.9. Hence, for boundary and core routers, the parameters are set as summarzed n Table 4.2. Bucket Sze (bts) 3e e+06 2e e+06 1e asterx smpsons bond 3vdeo asterx bond smpsons 3vdeo 0.2*x 0.2*x 0 0 1e+06 2e+06 3e+06 4e+06 5e+06 6e+06 Token rate (bps) Fg Vdeo characterzaton The characterzaton results obtaned consderng the consdered three vdeo flows (named Goldfnger, Asterx and Smpson), the aggregate of 15 voce sources (correspondng to a host n the smulaton scenaro) and the data traffc are reported n Table 4.1. In the table, the column Dmax ndcated the maxmum end-to-end delay analytcally obtaned from the estmaton of scheduler parameters. Traffc flow Rate (ρ) Buffer (b) (Kbt) Dmax (ρ/b) (msec) GOLDFINGER ASTERIX SIMPSONS VOICE sources Data Table Traffc characterzaton of consdered sources In the settng of the scheduler parameters, we can choce to set a servce rate equal to the sum of ρ obtaned for the three sources, and a buffer sze equal to the sum of b obtaned for each source. Consderng ths approach and the results obtaned wth the procedure presented above, the total servce rate to allocate for the vdeo servces and the related buffer sze are equal to 3.35 Mbps and 670 Kbts respectvely. The other approach conssts n the estmaton of the parameters drectly from the LBAP curve of the multplexed traffc. In ths case, t s expected to obtan a multplexng gan n the settng of the resources to guarantee to the vdeo servce. In partcular, consderng the same upper bound of the end-to-end delay, we need to allocate a servce rate of 2.5 Mbps and a buffer sze of 500 Kbts. Then, n terms of buffer sze we observe a gan of 170 Kbts (correspondng to a reducton of about 25%), whle n terms of servce rate a gan of 850 Kbps (25%) s acheved. Consderng the servce rate evaluated for each knd of traffc source, t s possble to Boundary Routers Output Servce Rate 3.06 Core router Core router Table 4.2 Parameters of the routers The buffer sze s gven for each traffc class,.e. voce, vdeo and data. When consderng the scenaro related to the aggregaton of voce and vdeo flows, the buffer sze for ths aggregated class of traffc has been set equal to B voce +B vdeo. 5. Smulaton results Buffer Sze, B (Kbt) B voce =30 B data =500 B vdeo =250 B voce =60 B data =1.000 B vdeo =500 B voce =90 B data =1.500 B vdeo =750 The smulaton analyss s manly focused n the evaluaton of the mpact of dfferent aggregaton strateges on the QOS parameters. The low scalabty of the IntServ network archtecture, suggests for the IP Telephony scenaro to study a DffServ core network archtecture. Hence, t s expected that n each node of an IP Telephony core network, a per-class queueng s mplemented and an approprate schedulng algorthm guarantees the QoS requested by the real tme sources. Two relevant problems arse n ths framework. The frst concerns the choce of an approprate schedulng algorthm and of a procedure for the settng of ther parameters. The second ssue s related to the aggregaton strateges to adopt. Hence, n ths Secton we present the results that gve nsghts on these problems. We consder two dfferent aggregaton strateges: n a frst one we have carred vdeo and voce together n a premum class and data n a best effort class (n the fgures the related curves are ndcated wth label contanng the word 1 lnk ); n a second one we have supposed to carry vdeo traffc n a separate queue from voce traffc (curves labeled wth the noun contanng the word 2 lnk ). Fg 5.1 presents the complementary probablty of the end-to-end delay expermented by the voce packets when the two dfferent strateges are consdered and only one vdeo source s actvated (n ths case the settng of smulaton parameters have been changed accordng to the dmensonng procedure presented n the prevous paragraph, n order to take nto account the nactvty of the others vdeo sources). Fg. 5.2 presents the same curves obtaned when all three vdeo sources are actve.

5 As frst analyss of the smulaton results, t s possble to note that the adopted schedulng algorthm,.e. WF 2 Q+, and the procedure for ts parameters settng permt to guarantee the target QoS for the voce sources f the vdeo and voce traffc flows aren t merged n a sngle queue. In partcular, the curves labeled as 1 Lnk ether n Fg. 5.1 and n Fg. 5.2 clearly show that the maxmum delay observed durng the smulaton s under 10 msec, whch s lower than the fxed delay of 27 msec, consdered n the settng of the scheduler parameters. Complementary probablty, P{Delay >τ} Complementary probablty, P{Delay >τ} Voce delay 1 lnk Voce delay 2 lnk e Delay, τ (sec) Fg Complementary Probablty of Voce Delay: P{delay > τ} - One Actve Vdeo Source 1 Voce delay 3 vdeo 2 lnks Voce delay 3 vdeo 1 lnk 1e Delay, τ (sec) Fg Complementary Probablty of Voce Delay: P{delay > τ} - Three Actve Vdeo Source Both Fg. 5.1 and Fg. 5.2 show the degradaton n terms of end-to-end delays of voce traffc when the vdeo flows s merged n the same queue wth the voce traffc. Indeed, n the Fg. 5.1, we can note that n correspondence of a probablty P=0.001 a delay of 7 msec s observed n the frst case (curve 1 Lnk ), whch s lower than the 15 msec regstered n the second case. Furthermore, ths degradaton s amplfed when the number of vdeo sources s ncreased. Indeed, n Fg. 5.2 the maxmum end-to-end delay regstered for voce traffc s unvared wth respect to the prevous case,.e. about 7 msec, whle t s ncreased to 23 msec, when all real-tme flows are aggregated n the same queue. Hence, n ths second case the worsenng of the delay parameter of voce servce s due to the ncrease of the number of bursty traffc multplexed wth the voce sources. Hence, we can suppose that f there s more requested bandwdth the need for network resources ncrease n a non-lnear way when consderng a wrong strategy of aggregaton, n ths case the real tme applcaton may be damaged serously. On the other hand, the vdeo performance take benefts from the aggregaton, showng a lttle delay mprovement that can be related to the multplexng wth the voce (the related fgures are not reported for sake of smplcty). Comp. Prob., P{Delay Jtter >J} Delay jtter 3 vdeo 2 lnks Delay jtter 3 vdeo 1 lnk 1e Delay jtter, J (sec) Fg Complementary Probablty of Voce Jtter Delay: P{ delay jtter > τ} - Three Actve Vdeo Source The dfferent performance observed wth the two consdered aggregaton strateges, can be related to lockout phenomenon, whch plays a decsve role n deteroratng voce performance. Indeed, when the vdeo sources are merged wth the voce traffc, the frst monopolze the queue space obstructng the second from recevng the desred servce level. The lock-out phenomenon can be avoded usng dfferent queues for vdeo and voce traffc, whle, at the same tme, the choce of approprate schedulng dscplnes can guarantees an adequate multplexng gan. Fnally, we observed that the best effort traffc (used as background traffc) s not affected by the fuson of the two servce classes because the total bandwdth share (vdeo + voce) s unchanged. The same worsenng of performance can be observed when we consder the jtter parameter, as shown n Fg. 5.3, whch plots the complementary probablty estmated for ths statstc (the results are related to the multplexng of three vdeo sources). 6. Concluson The man goal of the paper s the evaluaton of dfferent traffc aggregaton strateges for voce and vdeo servces n a Dffserv envronment. In ths framework, the smulaton analyss presented n the paper hghlghts that the wrong aggregaton of traffc flows wth dfferent

6 statstcal features, such as vdeo and voce traffc, may lead to performance worsenng, whch should be avoded especally n provdng IP-based busness servces, such as IP Telephony. On the other hand, the smulaton results emphasze that wth an adequate solaton between vdeo and voce traffc flows and an approprate dmensonng of network resources, t s possble to provde real-tme servces. In partcular, the consdered WF 2 Q+ schedulers network and the proposed procedure for settng the related parameters, permt to acheve the target QoS. Furthermore, analyzng the proposed procedure for the settng of schedulers parameters, based on the LBAP traffc characterzaton, t has been possble to hghlght the multplexng gan obtanable consderng the LBAP characterzaton of aggregated traffc. Fnally, the smulaton results have evdenced that although we have neglect the latency terms n the expresson of the end-to-end delay reported n lterature, the maxmum delay expermented s lower than that analytcally estmated (see the fourth column n Table 4.1). Ths result s a further evdence of the very conservatve nature of the worst-case analyss. Acknowledgment Ths work was partally carred out wth the support of the EU n the framework of the Project MOICANE, IST References [1] K. Dolzer, W. Payer, M. Eberspacher A Smulaton study on Traffc Aggregaton n Mult-Servce Networks, Conference on Hgh Performance Swtchng & Routng (jont IEEE ATM Workshop 2000), Hedelberg, Germany, June, 2000 [2] K. Dolzer, W. Payer On aggregaton strateges for multmeda traffc Proceedng of the frst Polsh-German Teletraffc Symposum, Dresden, September 2000 [3] S. Blake, D. Blake, M. Carlson, E. Daves, Z. Wang, W. Wess An archtecture for Dfferentated Servces, Internet RFC 2475, December, 1998 [4] H. Naser, A. Garca, O. Aboul-Magd Voce over Dfferentated Servces, Internet Draft, Dffserv Workng Group, December, 1998 [5] S. Keshav An Engneerng Approach to Computer Networkng, Addson-Wesley, January, 1998 [6] J. Bennet, H. Zhang WF2Q: Worst-case Far Weghted Far Queueng, Proc. Of IEEE Infocom 96, March, 1996 [7] J. N. Dagle, J. D. Langford "Models for Analyss of packet Voce Communcatons Systems", IEEE JSAC, Vol. 6, pp , 1986 [8] O.Rose Statstcal propertes of MPEG vdeo traffc and ther mpact on traffc modelng n ATM systems, Un.of Wuerzburg - Inst. Of Computer Scence Research Report Seres. Report N February 1995 [9] R.G. Garroppo, S. Gordano, M. Pagano, G. Procss, "On the Relevance of Correlaton Dependences n On/Off Characterzaton of Broadband Traffc", Proc. of IEEE ICC 2000, New Orleans, Lousana, USA, June, 2000 [10] D. Stlads, A. Varma, Latency-Rate Servers: A general model for analyss of traffc schedulng algorthms, Tech. Rep. UCSC-CRL-95-38, July 1995 [11] A. Charny, F. Baker et al. EF PHB Redefned Internet Draft, 01.txt, November, 2000 [12] R. Bruno, R.G. Garroppo, S. Gordano "Estmaton of token bucket parameters of VoIP traffc", Proc. of IEEE ATM Workshop 2000, Hedelberg, Germany, June, 2000