On Signaling Efficiency for Call Setup in all-ip Wireless Networks

Transcription

1 O Sigalig Efficiecy for Call Setup i all-ip Wireless Networks Miguel Melyk ad Admela Juka Uiversity of Illiois at Urbaa Champaig {mmelyk2, juka}@uiuc.edu Abstract This paper aalyzes the sigalig efficiecy for multimedia call (sessio) establishmet i all-ip wireless etworkig ifrastructures based o the Iteret Multimedia Subsystem (IMS) stadard framework ad CDMA2000 Evolutio Data Oly Rev. A (EV-DOrA) stadard for wireless lik trasmissio. We preset a comprehesive, bottom up aalysis of sigalig delay for settig up multimedia sessios ad we evaluate system architecture alteratives to reduce it. Call setup time is particularly critical for iteractive applicatios, such as gamig, push-to-x ad Voice over IP (VoIP), where user experiece of time that takes to start or joi a sessio directly traslates i user perceptio of service quality. Our aalysis takes ito accout a large set of system desig parameters across all layers, ragig from radio lik properties to IMS processig ad specific characteristics of the Sessio Iitiatio Protocol (SIP). For a example sigalig flow betwee two mobile users, we derive a importat set of umerical results for call set up time uder various IMS desig scearios that cosider effects of SIP compressio efficiecy, choice of trasport protocols (TCP, UDP) ad radio lik quality. Based o the results obtaied, we propose ad discuss ew methods to improve the sigalig efficiecy ad to reduce the call setup time. 1. Itroductio Emergig mobile ad wireless applicatios - such as multiparty video sessios, iteractive etertaimet, multimedia istat messagig, ad push-to-talk commuicatio - are drivig the covergece of wireless ad mobile commuicatios, Iteret Protocol (IP), ad distributed data services. I this cotext, a umber of stadards are beig developed aimig at addressig the desig of commo platforms for service creatio ad provisioig, as well as of the related sigalig protocols (e.g., Sessio Iitiatio Protocol, SIP). The most recet example is the Iteret Multimedia Subsystem (IMS), a stadard used to desig a advaced service platform that ca provide a wide rage of mobile, multimedia applicatios o IP ifrastructure. IP Multimedia Subsystem (IMS), defied by the 3 rd Geeratio Partership Projects (3GPP, 3GPP2), is emergig as a defacto stadard to desig commo platforms for service creatio, provisioig ad delivery [2]. It has bee origially desiged to itegrate cellular ad IP etworks, but its broader scope is i geeric service platform for rapid deploymet of ew applicatios. IMS ca provide a scalable service platform desig for existig ad emergig services, such as mobile wireless services, Iteret services ad services developed dyamically. The service examples rage from push-to-talk over cellular, real-time multimedia services to iteractive etertaimet ad voice ad istat messagig. IMS is desiged to operate idepedetly of the type of access devices (statioary, mobile, wired, wireless). IMS uses the uderlyig Iteret etworkig ifrastructure ( all IP ), a choice made due to the ever-growig data traffic over the traditioal voice etworks (e.g. wireless cellular) ad due to the domiace of the IP etworkig techologies. With wide acceptace of Voice over IP (VoIP) ad expectatio for rapid adoptio of multimedia cellular services, the sigalig efficiecy i IMS has become icreasigly importat to the operators. Users ot oly expect high voice ad video quality, but also high availability, speed ad reliability i service ivocatio ad executio. I this cotext, the call set up time critical to real-time ad iteractive applicatios is becomig oe of the key parameters of Quality-of-Service (QoS). For example, VoIP user dial time is expected to be comparable to that of telephoy etworks, ad to joi a ogoig gamig sessio should take just a secod. IMS uses SIP to perform support fuctios of user registratio, user locatio ad presece iformatio, autheticatio i support of mobility, sessio set-up ad maagemet, etc. I curret IMS ifrastructures that use text-based SIP, sigalig efficiecy has become a importat optimizatio objective for call setup, particularly i the presece of lossy ad capacity costraied wireless liks. Motivated by the importace of sigalig efficiecy for improved call (sessio) setup time, i this paper we comprehesively aalyze the specific issue of SIP sessio set up time i IMS ifrastructures. We model for the first time to our kowledge the sigalig message flow ed-to-ed ad bottom-up, icludig wireless trasmissio, IMS cotrol elemets ad ed-users mobile devices. Our model is based o realistic assumptios for the system desig parameters, icludig radio lik delays i EVDOrA systems, IMS processig properties ad SIP messagig. By profilig the sigalig properties ad mappig them to the IMS ad to wireless lik fuctios, we derive importat results for sessio set up time as a fuctio of SIP message compressio, trasport protocols (TCP, UDP) for sigalig ad radio trasmissio quality. The paper is orgaized as follows. Sectio 2 briefly describes the IMS ad explais the SIP message flows used. Sectio 3 presets the aalytical model of the ed-to-ed sigalig procedures. Sectio 4 defies the model evaluatio /06/$20.00 (c) 2006 IEEE

2 scearios ad aalyzes the umerical results. Sectio 5 cocludes the paper. 2. The IMS model ad SIP messages flow I this paper, we focus o sigalig efficiecy related to the geeric multimedia sessio setup betwee two users (MT1 ad MT2), who are subscribers of differet operators. The multimedia sessio cosists of a two-way real-time RTP/RTCP (Real Time Protocol) sessio. Without loss of geerality, we assume that users are attached to their home etworks (i.e. o roamig). Similar scearios ca be easily derived for the roamig users. Figure 1 illustrates a high-level view of ed-toed IMS architecture ad etwork ifrastructure used i this paper to model the call (sessio) setup delay. We assume that both carriers A ad B use CDMA2000 wireless trasport, ad have idetical architectures i their respective domais: radio access etwork or backhaul (BTS, BTC, PCF, PDSN compoets, itercoected with poit-to-poit liks) ad a highspeed IP backboe. I geeral, the basic buildig blocks i the IMS architecture iclude several compoets or fuctios: CSCF (Call Sessio Cotrol Fuctio), HSS (Home Subscriber Server), a umber of gateway fuctios (Media, Breakout, Sigalig), service ad applicatios elemets (AS). 1 There are three types of CSCF cotrol elemets: Proxy (P-CSCF), Iterrogatig (I- CSCF) ad Servig (S-CSCF). P-CSCF is the etry poit for radio access etwork to IMS ad is resposible for all commuicatio with the home etwork. I-CSCF is the first poit of cotact i the home etworks ad it also commuicates with the HSS. Fially, S-CSCF is the mai poit of service ad provides full sessio cotrol ad autheticatio. Without goig ito further details, it is importat to ote that a umber of iterfaces ad protocols are defied to eable commuicatios amog these compoets; Cx iterface eables commuicatio betwee I (or S)-CSCF ad HSS, Sh iterface for AS ad HSS, etc. Amog all protocols used i IMS (e.g., IP, TDM, SS7 ad Diameter), SIP is the oe that provides most of the fuctioality for sessio iitiatio ad maiteace, as well as user locatio ad availability fuctios. For geerality, we assume that oly P-CSCF is deployed ear the PDSN, while the remaiig IMS compoets are distributed withi the high-speed IP backboe. The detailed message flow aalyzed is show i Fig. 2. It follows a typical IMS call setup exchage [2, 5], i which SIP messages are routed through the differet CSCF elemets, ad trasferred betwee operators as defied by the IMS stadard. We assume that both termials are already registered by the time the sessio establishmet process begis, for which the user is already autheticated i the IMS ad assiged to P- CSCF ad S-CSCF. That icludes the commuicatio betwee S-CSCF ad HSS regardig the user profile. The call setup process accouts for the RTCP message exchage. At least oe RTCP packet carryig the CNAME of the other termial eeds to be received before the RTP stream ca start [3]. The sessio setup starts whe the caller seds a message (1). The P-CSCF checks the message formatio for correctess ad forwards the message to the assiged S-CSCF. By recordig their correspodig addressig iformatio i the Route SIP header field, the P-CSCF ad S-CSCF i both operators are always ivolved durig the etire setup process. The S-CSCF i the home etwork discovers the I-CSCF i the termiatig etwork by meas of DNS queries (ot cosidered i our aalysis) ad forwards the message. The I- CSCF queries the HSS to discover the S-CSCF assiged to the callee, ad forwards the message to it. The message reaches the termial through the P-CSCF assiged to it, after iitial filterig is performed at the S-CSCF. This message flow assumes that a QoS precoditio is set [5], for which all SIP messages i betwee the origial IN- VITE (1) ad the message (8) are related to commuicatig the QoS egotiatio status to the other ed. The messages related to this egotiatio are ot show i the diagram, we assume that they do ot add delay to the call setup process. The callee is oly alerted whe the resource reservatio process is fiished at both eds. Whe the call is aswered, the callee issues the related to the origial. Whe received, the caller starts RTP/RTCP media trasmissio ad geerates the fial message that cocludes the trasactio. The receiver starts media RTP/RTCP media trasmissio upo receivig the ad the call is established. The detailed aalysis of the SIP messagig process is out of the scope of this paper. However, it is worth otig that, while the show flow is simplified, it takes ito cosideratio all elemets of sigalig that typically cotribute to the call setup time. For istace, whe SIP is delivered over TCP, the TCP coectio establishmet hadshake is accouted for. 3. Aalysis To study the ed-to-ed call (sessio) setup delay, we performed a mea value aalysis by modelig ad aggregatig the delay of all packet trasmissio ad IMS processig evets for the sigalig flow show i Fig 2. The followig subsectios describe i detail the delay aalysis of each evet ad compoet uder cosideratio. The preseted model is similar to the aalysis of the related protocols, e.g., H.323 as i [3], but it icludes ovel elemets i several aspects. First, it cosiders two mobile ed-poits i multiple operators. Typically, oly oe mobile user is cosidered i a sigle operator sceario. Secod, we model the Radio Lik Protocol (RLP) ad the packet trasmissio delay based o the stadard physical layer ad protocol specificatios of EVDOrA [1]. Third, we propose methods for iter-layer optimizatio, such as SIP compressio that fits i oe radio lik frame. We believe that this is oe of the earliest attempts to comprehesively study the sigalig efficiecy from physical to applicatio sessio layer, ad their iteractios. 1 For the detailed descriptio of the IMS architecture ad protocols refer to [2,5]. 1940

3 Operator A Operator B Mobile Termial 1 P-CSCF I-CSCF I-CSCF P-CSCF Mobile Termial 2 BTS BSC PCF PSDN IP backboe A IP backboe B PDSN PCF BSC BTS HSS HSS RAN Backhaul S-CSCF S-CSCF AAA AAA Figure 1. IMS architecture with CDMA2000 EVDOrA used i this paper. MT 1 P-CSCF S-CSCF Itercoect I-CSCF HSS S-CSCF P-CSCF MT 2 Dial Ed (1) 183 S.P. (4) UPDATE (6) (10) 183 S.P. Operator A Operator B Iteral IMS Processig Diameter LIR Diameter LIA (Iitial Filter) UPDATE UPDATE UPDATE (Iitial Filter) RTCP SDES RTCP SDES (11) MEDIA Figure 2. IMS sigalig flow aalyzed i this paper. (9) HSS Database Query (2) (3) (5) UPDATE (7) (8) Rig Aswer 3.1 Radio Lik Sigalig Delay (RLD) Oe of the most importat variables i our wireless lik aalysis is the Frame Erasure Rate (FER), defied as the ratio of errored frames discarded by the receiver over total frames set. The FER represets a measure of the istataeous quality of the radio lik at the time the packets uder study are set, ad is used thoroughly i our radio lik models. The total delay experieced by a packet across the radio lik is impacted by several factors, icludig the physical layer characteristics of the wireless lik, the schedulig policy, the retrasmissio fuctios i RLP ad the trasport protocol used o top of the CDMA2000 system. The RLD model cosiders the physical layers of EVDOrA [1] for the Forward Lik (FL) ad the Reverse Lik (RL) separately. To model the FL, we assumed that oly sigle-user MAC frames are used, ad cosidered the effect of early termiatio due to the Hybrid Automatic Request (HARQ) algorithm. HARQ will cause that some frames with a omial spa of k slots fiish trasmissio earlier, usig effectively k slots istead. We cosider i our model the effect of early termiatio by upper-boudig k, ad usig k istead of k for estimatig the trasmissio delay. This upper boud was estimated usig field measuremets of early termiatio gais [8]. The value of k used for each Data Rate Cotrol (DRC) is show i Table 1. DRC k [slots] k [slots] , ,7,8, ,11,12, Table 1. Nomial Spa ad Early Termiatio [8]. Due to the 4-slot iterlacig scheme, a physical layer frame with omial spa k slots will have a trasmissio delay D = 4(k -1)+1 slots (a EV-DOrA slot is 1.66 ms log for both FL ad RL [1]). Sice the FL is shared by all users i sector, each frame will experiece queuig delay i additio to the trasmissio delay. We made the simplifyig assumptio that each radio frame i the stream will experiece a fixed iter-frame queuig delay D q. The total trasmissio delay as experieced by the upper layers id D=D +D q. Uder this assumptio, the operatio of the scheduler is trasparet to timers ad retrasmissio mechaisms i the upper layers. It is merely 1941

4 see as a icrease i the total trasmissio delay. The queuig delay D q is variable accordig to the sector FL load. To obtai the value of D q we have simulated the FL operatio. Cosiderig that a sector has a omial rate of 600 slots/s, we defied a simplified traffic model with 50 active termials (AT) i the sector. Each AT is coected with a DRC [8]. Packets goig to each AT arrive to the scheduler accordig to a Poisso process with idetical arrival rate, ad the FL schedules trasmissio usig FIFO queuig disciplie. The aggregated offered load i slots per secod A cosiderig all DRC categories i is: A = N i k' i i where N i is the umber of ATs coected with DRC. The simulatio results (skipped here for clarity) show that for a slot load of 80% of the maximum load, D q mea value is 15 ms, ad the 95 percetile value is 50 ms. To accout for burstiess, we cosidered 30 ms as a adequate fixed value to use i our model. To model the RL, we cosidered the effect of the HARQ algorithm i the same way as for the FL. If k is the target umber of sub-frames ad k is the effective umber of subframes due to HARQ, we assume k =k-1 for k>1 ad k =1 if k=1. Due to the 3 sub-frame iterlacig scheme, a physical layer frame will have a trasmissio delay D = 4(3(k -1)+1) slots. For RL, the queuig delay is assumed egligible. 3.2 RLP Sigalig Delay We derived a probabilistic model to study the delay of RLP based o the detailed EVDOrA stadard specificatio i [1] ad the model proposed i [3]. Our model takes as iput the physical layer FER ad the physical layer frame trasmissio delay D to estimate the loss rate ad trasmissio delay of a trasport-level packet usig RLP. For our RLP model, we assume that: A trasport-level packet is received at the EVDOrA applicatio layer, which will divide the packet i frames accordig to the allowed physical layer frame payload size give the curret lik coditio. The RLP protocol will sed each frame immediately after the trasmissio of the previous oe eds. No other traffic sources to the same AT are active. The receiver will sed 1 RLP N for each frame that is detected lost. The probability p of frame error (FER) is equal i both the FL ad RL. Uder these assumptios, ad cosiderig that a trasportlevel packet is trasmitted successfully if all its related datacarryig radio frames are trasmitted successfully (either i the first attempt or i the retrasmissio), the probability that a sigle RLP frame is trasmitted successfully is: 2 PS RLP = 1 p (2 p) The probability of successful packet trasmissio is the: PS = PS RLP Defiig j as the umber of frames iitially lost ad the recovered by retrasmissio, we compute D RLP (the average delay of a trasport-level packet over RLP) for the case of successful packet trasmissio. To compute D RLP uder these assumptios, we cosider the evets that geerate Ns i the receiver. To detect packet loss, the receiver eeds subsequet packets to compare the sequece umbers of the ew ad previous bytes. Oce the loss is detected, Ns are geerated. If o ew applicatio frames are set, the receiver will lear about the loss oly whe the seder s flush timer expires ad a probe packet cotaiig at least the last byte previously set is geerated. This situatio will occur if the last oe of the frames belogig to this TCP packet is lost. The average RLP delay eeds to cosider the effect of losig the last frame, for which we computed the coditioal expectatio D RLP give ad j. Defiig L as the evet i which the last frame has bee lost i the trasmissio we have: P(L, j ) = 1 j 1 = j j Besides, P(L, j=0)=0 ad P(L =1, j=1)=1 (>1, j >1) For the sake of simplicity, we also assumed that Ns are delivered istatly to the seder. If D is the frame trasmissio delay, ad T s is the flush timer expiratio time, for a sequece of frames: j j E( DRLP, j) = ( + j) D + (( + j) D + Ts ) Assumig that the probe packet is always received successfully, D RLP is calculated as: D RLP = 1 (1 p) j ( p(1 p) 2 ) j E(D, j) PS j j= Trasport Protocol Sigalig Delay IMS requires reliable trasport of SIP packets. We cosidered for our aalysis 2 trasport optios: TCP ad UDP. The upper layers see the wireless coectio as a chael with oe-way delay D RLP ad a packet loss rate q=1-ps. Depedig o the trasport protocol i use, the applicatio layer may experiece differet average delays, maily because of the way the trasport layer hadles timeouts ad retrasmissios. I ay case, a retrasmissio time-out (RTO) parameter is required. A packet is cosidered lost by the seder if o ackowledgemet is received before the retrasmissio timer, origially set to RTO secods, expires. TCP has a welldefied stadard retrasmissio algorithm to estimate the roud-trip time (RTT), origially thought for traditioal etworks, ad defies a retrasmissio time-out based o this value. O the other had, if UDP is used as SIP trasport, the applicatio layer must take charge of coductig reliable commuicatio ad also of estimatig the RTT ad RTO. I ay case, the estimatio of RTT should coverge quickly ad follow closely the actual RTT. Ideally, RTO should be equal to RTT (the miimum possible time ad ackowledgemet ca be received after a packet was set) TCP Model We cosider the stadard TCP RTO computatio ad retrasmissio with expoetial back-off. The probability dis- 1942

5 tributio of the umber of attempts required to successfully trasmit a TCP frame is geometric with parameter q, ad for a total of 3 trasmissio attempts the average delay is: 1 2 i i DTCP = (1 q) q ( D (2 1) RTO) 3 RLP + 1 q i= 0 The value of RTO is updated every time a packet successfully trasmitted is ackowledged. RTO is iitially set to 3 secods, ad ca have a miimum value of 1 secod. RTO is estimated based o the smoothed RTT ad the mea RTT deviatio as idicated i [6], ad it typically takes several successful packet trasmissios to coverge to a steady-state value. Sice the etire IMS call takes oly 6 SIP messages to fialize, the RTO will typically be i its trasiet period throughout the call. For simplicity, however, we assume that RTO remais costat throughout the IMS call [7]. This costat value of RTO is estimated as 2 secods for our study, give that a TCP packet of average size 600 bytes ca have a 1 secod RTT o the EVDOrA lik UDP Model I this case, the SIP applicatio is resposible for hadlig retrasmissios. We assume that the SIP applicatio will do up to 3 trasmissio attempts. As idicated i [4] ad [10], we cosider that the SIP applicatio follows the expoetial backoff retrasmissio algorithm, estimates RTT ad RTO, ad stops retrasmissio attempts upo the receptio of a provisioal (i.e. 100 Tryig) or defiitive respose (i.e. ). Uder these assumptios, the delay estimatio for UDP is idetical to that of TCP, the oly differece beig the estimatio of RTO ad the coectio establishmet hadshake (abset i the UDP case). The delay for a UDP packet is estimated i the same way as for TCP. 3.4 IP Backboe ad Backhaul Delays To develop realistic estimates of IP backboe delays we referred to actual delay measuremets i real implemetatios. I [9], a 28 ms oe-way delay was measured betwee the East ad the West Cost of the US. I our model, we assume the followig: All IMS fuctios i a operator domai, as well as the operators itercoect poit, are distributed withi the IP backboe. The trasmissio delay betwee ay of these backboe poits is 15 ms. The P-CSCF is deployed close to the PDSN so that backboe delay is egligible betwee them. Iter-domai trasmissios always icur 35 ms of delay. Cocerig the delay i the CDMA2000 packet etwork (backhaul) we assume that PDSN, PCF, BTC ad BTS are coected over 1.5 Mbps poit-to-poit liks. The utilizatio of these liks is 60% ad the average packet size trasmitted is 500 bytes (trasmissio time of 2.66 ms over 1.5 Mbps). By usig a simple M/M/1 queue model, we derive the mea value of 20 ms as the total PDSN-to-BTS delay. 3.5 Processig Delays To characterize the impact of the seemigly log route that SIP messages eed to trasit at the applicatio level, we also cosider the processig time spet i IMS processig at the CSCF elemets ad i queryig of HSS for S-CSCF locatio. Cocerig the SIP processig time, the typical latecies i curret IMS/SIP commercial solutios are less tha 50ms. We used 30ms for this aalysis. The HSS latecy is typically less tha 300ms for Diameter protocol requests. 3.6 Impact of SIP Packet Size SIP is a text-based protocol which ca ecapsulate differet payload types, or other text-based protocols. The size of a typical SIP message ca rage from several hudreds of bytes to a few thousad bytes. To obtai typical SIP message sizes for our aalysis, we used the sample SIP messages i [5]. We also cosidered that SIP messages ca grow as they are routed across multiple SIP servers (Record Route ad other fields). The iheret redudacy of SIP messages as a text-based protocol leads to iefficiet call setup delays, particularly i low rate, lossy liks. For this reaso, SigComp [11] was developed by the IETF for geeral text-based protocol compressio. IMS stadardized SigComp as of madatory implemetatio i the P-CSCF (Proxy) ad the UA (User Aget). SigComp specifies the use of the Uiversal Decompressor Virtual Machie (UVDM) at the receiver, which allows the use of ay compressio algorithm [11]. For the purpose of uderstadig ed-to-ed call setup sigalig delays, it is importat to uderstad the actual compressio efficiecy of SigComp. The experimets i [12,13] report the compressio performace iformatio as they test actual SIP flows with SigComp compressio i differet scearios. From these results we ca observe that the compressio rates ca be cosidered as 40-60% for the iitial message (), ad depedig o the particular message type, 65% to 85% for other subsequet messages i the SIP flow. For our aalysis, we chose 50% for the messages ad 75% for subsequet messages. Table 2 shows the ucompressed ad compressed SIP message sizes usig this estimated compressio rates. Message Type Ucompressed Size [byte] SigComp Size [byte] (1) 1, (2) 2,400 1, (S. P.) (3) 1, (4) 1, (5) 1, UPDATE (6) 1, (7) 1, SIP (8) SIP (10) RTCP SDES (9,11) 100 N/A Table 2. Size of IMS messages used i the model. 4. Sigalig Scearios ad Results I order to gai isights o how the IMS sessio setup delay is geerated ad how delay reductio ca be achieved, we modeled four scearios. All scearios assume the sigalig flow show i Fig. 2 ad also assume that SIP messages are 1943

6 compressed usig Sigcomp. We first defie the basic referece sceario (BASELINE) with TCP trasport. For the three remaiig scearios, we chage oe or more key architectural aspects impactig call setup sigalig delay, such as size of SIP messages, degree of cetralizatio of the IMS implemetatio ad trasport protocols utilized over the wireless liks. The followig scearios are cosidered: BASELINE-IMS: This is a referece sceario usig the TCP trasport over the radio liks. SLIM-SIP: This sceario assumes that all SIP messages are additioally ecoded so that all the iformatio coveyed i oe SIP packet fits i 1 sigle radio frame ( iterlayer optimizatio). This hypothetical assumptio refers to the best possible SIP message compressio case. PEER-SIP: This sceario is the same as SLIM-SIP, but assumes i additio that P-CSCF is able to act idepedetly of the other IMS compoets i the domai A ad ca establish a sessio directly with its peer P-CSCF i Operator B. This case avoids cetralized processig, while maitaiig the Operator s cotrol over accessed services. PEER-UDP: This sceario is the same as PEER-SIP, but uses i additio a ad-hoc reliable trasport protocol based o UDP betwee the termial ad the P-CSCF. We assume that each ed is aware of the wireless lik state i a way the exact estimatio of RTT ad RTO is possible (we use the hypothetical assumptio RTO=RTT). This sceario is the best possible case of reliable trasport, sice it is highly tailored to the uderlyig physical lik. To uderstad how the quality of the wireless coectios impacts the delay i a IMS call set up, the above scearios were aalyzed usig 2 differet wireless lik quality coditios based o the defiitios i [1]: Poor Coectio (PC): I the FL, we cosider the trasmissio format (1024, 16, 1024) associated with DRC 1, which uses the maximum slot spa i the FL (16) ad a low payload size of 1024 bits. I the RL lik, we cosidered a 1024 bits payload. Good Coectio (GC): We cosidered the format (5120,1,64) (DRC 14 i the FL), ad a 6144 bit payload over 1 sub-frame i the RL. For simplicity, we selected trasmissio formats that provide a similar payload size i both the FL ad RL so the same umber of radio frames is used i both directios. These coditios apply simultaeously to both ATs i Figure 1. Table 3 shows the umber of resultig radio frames whe trasmittig each compressed applicatio message type over Good (GC) ad Poor (PC) wireless coectios. The ed-to ed call setup time was computed as the sum of the delays caused by all sigalig packet trasmissios ad all processig evets, as described earlier. We break dow the sessio setup i 2 phases as described i [10]: Dial-to-Rig Delay (D-to-R): From the call iitiatio evet (Dial) to the time the callee is alerted (Rig). Aswer-Sigal Delay (A-to-S): From the time the callee aswers to the time the media flow starts. Message Type Size [bytes] # Frames (PC) # Frames (GC) (1) (2) 1, (S. P.) (3) (4) (5) UPDATE (6) (7) (8) SIP (10) SDES (9,11) Table 3. Size of IMS Messages Used i the Model Figures 3-6 show the umerical results for the sessio setup times obtaied with our aalytical model. They show each call setup phase, i Poor ad Good wireless lik coditio. The call setup delay is plotted for differet values of FER. Based o the results obtaied, the followig observatios ca be made for the Poor Coectio (PC) case: I BASELINE-IMS, the etire dialog is traslated ito a fairly large umber of small-payload radio frames, each of which takig several tes of EV-DOrA slots to trasmit. As a result, we see a large D-to-R delay with high sesitivity to frame loss due to icreased probability of timeout ad retrasmissio. SLIM-SIP assumes cross-layer optimizatio betwee radio frames ad SIP compressio, ad reduces the umber of radio frames to the miimum possible; a average of 40% of setup delay compared to the BASELINE for equal radio lik quality. This shows the sigificat impact of the size of SIP messages o the overall setup time. As PEER-SIP avoids the cetralized authorizatio steps ad the origiatig P-CSCF proceeds directly to establish the call with its peer i the other domai, we see a combied improvemet of at least 60% with respect to BASE- LINE-SIP. PEER-UDP reduces the setup time further by elimiatig TCP hadshakes, with a overall setup delay reductio of 75% i D-to-R. I to A-to-S case, PEER-UDP ad PEER- SIP are idetical for low FER, give that the TCP hadshake already occurred i the D-to-R phase. For large FER, however, we ca see that PEER-UDP performs better i A-to-S due to the ideal RTO cosidered i this sceario. I the case of GC, the improvemet itroduced by SLIM- SIP is margial. Because of the larger frame payloads, most of SIP messages fit already i 1 frame. The largest improvemet i this case is achieved i PEER-SIP, by elimiatig the cetral IMS processig that takes about oe half of the total delay i the D-to-R phase. PEER-UDP provides also a additioal 10% to 20% reductio over PEER-SIP i the D-to-R phase, ad has idetical results as PEER-SIP i the A-to-S phase for low FER. 1944

7 Figure 3. Dial-To-Rig Delay (Poor Coectio). Figure 4. Aswer-Sigal Delay (Poor Coectio). Figure 5. Dial-to-Rig Delay (Good Coectio). Figure 6. Aswer-Sigal Delay (Good Coectio). As a fial remark, the call setup delays of PEER-UDP i both coectio coditios are remarkably similar. Despite the large differece i trasmissio rates ad radio frame delays, both scearios seem to operate close to a limit beyod which additioal improvemet seems to be difficult to obtai just by improvig the radio lik quality. I PEER-UDP the key delay-geeratig factors are the IP packet trasmissio, the EV-DO scheduler queue, ad the SIP processig delays. The ed-to-ed backboe trasmissio delay cosidered i this study is 35 ms, very close to the measured coast-to-coast delay of 28 ms [9]. The remaiig of these parameters is cosidered as fixed values i our model, reaso why we caot explore further reductios at this poit. However, all of them ca be reduced by etwork or applicatio desig (e.g., larger liks, less load per sector, faster SIP processig). 5. Coclusio I this paper, we preseted a aalytical model for evaluatig the call setup sigalig efficiecy i IMS ifrastructures usig CDMA2000 EV-DOrA radio lik trasmissio. Our model is based o realistic system parameter assumptios for radio lik delays, IMS processig, SIP sigalig ad SIP compressio. We showed prelimiary, but importat ad idicative umerical results ad preseted methods to improve the sigalig efficiecy ad hece reduce the call setup time. The use of the architecture alteratives to the basic IMS desig sceario seems to provide a bouded ed-to-ed call setup delay, which, ideally, is close to beig idepedet of the radio lik quality. We believe that the results we obtaied show importat directios for the IMS architecture ad its implemetatio. Our future work will cosider the validatio of our model through experimets ad the exploratio further iterlayer optimizatio. 6. Refereces [1] 3GPP2, cdma2000 High Rate Packet Data Air Iterface, C.S0024-A, March [2] 3GPP2, All-IP Core Network Multimedia Domai - IMS Stage 2, X.S , Dec [3] Das, S. et al., Performace Optimizatio of VoIP Calls over Usig H.323 Protocol, INFOCOM [4] Roseberg, J. et al., SIP: Sessio Iitiatio Protocol, IETF RFC 3261, Jue [5] Camarillo, G., Garcia-Marti, M, The 3G IP Multimedia Subsystem, Wiley, [6] Paxso, V., Computig TCP's Retrasmissio Timer, IETF RFC 2988, November [7] Mellia, M. et al., TCP Model for Short-Lived Flows, IEEE Comm. Letters Vol. 6 No 2, Feb [8] Q. Bi et al, A Forward Lik Perf. Study of the 1xEV-DO System Through Simulatios ad Measuremets, Bell Labs Techical Joural, March [9] Boutremas, C., Iaacoe, G, Doit, C., Impact of Lik Failures o VoIP Performace. [10] Curcio, I., Luda, M., SIP Call Setup Delay i 3G Networks. IEEE ISCC 02. [11] Price, R. et al, Sigalig Compressio (SigComp), IETF RFC 3320, Jauary [12] Fridrich, M., Bilicki, V., et al, SIP Compressio, Hugaria Academy of Scieces, May [13] Yi, S. et al, Post-Dialig Delay of Multimedia Sessios i 3G Networks, ICOIN 2004, Feb