Equation Based Rate Control and Multiple Connections for Adaptive Video Streaming in Cellular Networks

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1 Equaton Based Rate Control and Multple Connectons for Adaptve Vdeo Streang n Cellular Networks A. Mardanan Dehkord and V. Tabataba Vakl Abstract Rate control s an portant ssue for vdeo streang n cellular networks. In ths paper, we propose an equaton based rate control and ultple connectons for adaptve vdeo streang n cellular networks. In our ethod the sendng rate s calculated as a functon of round trp te (RTT), loss event rate (p), packet sze (s) and new control paraeters α and β, that are able to provde flexble and sooth transsson rate and slowly responsble congeston control and also adaptablty to unpredctable wreless channel condtons. On the other hand, by usng one TFRC connecton, the wreless bandwdth s underutlzaton, so we ntroduce a ethod by usng ore TFRC connectons wth new equaton, t has the potental to acheve optal perforance, axu throughput, and nu packet loss rate. We have sulated ths ethod n UMTS and accordng to results, ths ethod n addton to network stablty ncreases throughput wth low fluctuaton by varyng α and β and openng approprate nuber of connectons. Index Ters adaptve vdeo streang, cellular networks, rate control, TFRC, UMTS. I. INTRODUCTION As councatons technology s beng developed, user s deand for ulteda servces rases. One of the ost prosng servces s the transsson of rch ulteda content, but bandwdth s a valuable and lted resource for UMTS and every wreless network. So currently, the poor vdeo qualty on the low and fluctuated bandwdth networks s akng the use of streang applcatons very dffcult [], []. Congeston control s aed at solvng ths proble by adaptng the streang rate to the network condtons. A wdely popular rate control schee over wred networks s equaton-based rate control also known as TCP frendly rate control (TFRC). It explctly desgned for best-effort uncast ulteda traffc, but t s not a full transport layer protocol [3]. Soe applcatons requre a fxed nterval of te between packets and vary ther segent sze nstead of ther packet rate n response to congeston and accordng to [3] TFRC s not desgned for those applcatons. On the other hand, by usng the TFRC connecton n wreless streang applcatons results n underutlzaton of the wreless A. Mardanan Dehkord s wth Iran Unversty of scence and Technology, (correspondng author to provde phone: (+98) ; (e-al: a_ardanan@vu.ust.ac.r). V. Tabataba Vakl s wth Iran Unversty of scence and Technology, (e-al: vakly@ust.ac.r). 533 bandwdth. In ths work we ntroduce a ethod that s based on new throughput equaton and provdes flexble and sooth transsson rate, slowly responsble congeston control and also t has the potental to fully utlze the wreless bandwdth provded nuber of connectons. We have sulated ths ethod n UMTS and accordng to results, ths ethod n addton to network stablty ncreases throughput wth low fluctuaton. Ths paper s structured as follows. Secton s a descrpton of the related works. In Secton 3 the TFRC echans wth new equaton for the UMTS s evaluated. In secton 4 we propose strategy of usng ore connectons to acheve axu throughput. Secton 5 s dedcated to the analytc and sulaton results, Secton 6 presents concluson. Soe deas for future work are also outlned. II. RELATED WORKS Rate control s an portant ssue n both wred and wreless streang applcatons. References [] and [], present streang vdeo over UMTS transport channels through the use of the odfed the TFRC echans that s anly used n wred networks. Reference [4] presents, TCP Reno, treats the occurrence of packet loss as a anfestaton of network congeston. Equvalently, TCP Vegas uses queung delay as a easure of congeston [5]. Thus, the authors [4], propose an enhanceent of the TCP Reno and TCP Vegas for the wreless networks, naely TCP Veno. In Reference [6], two algorths are presented that forulate resource allocaton n wreless networks. In Reference [7], the perforance characterstcs of TCP New Reno, TCP SACK, TCP Veno and TCP Westwood under the wreless network condtons are studed and the authors propose TCP New Jersey, whch s capable of dstngushng wreless packet losses fro congeston. In other schee [8], when a packet s lost, TFRC goes beyond layerng abstracton and enqures the lnk layer about the recent sgnal strength. The packet loss s recognzed to be due to wreless channel error f recent sgnal strength s low and due to congeston otherwse. Explct Loss Notfcaton (ELN) can also be appled to notfy TCP/TFRC sender when a packet loss s caused by wreless channel errors rather than congeston [9]. In Reference [0], the authors nterpret loss as a sgn of congeston f one-way delays are ncreasng and a sgn of wreless channel error otherwse. Slarly, Baran and Matta [] proposed a loss dfferentaton schee based on the assupton that the varance of round trp te s hgh when congeston occurs,

2 and s low otherwse. Cen et. al. present an end-to-end based approach to facltate streang over wreless []. They cobne packet nter-arrval tes and relatve one way delay to dfferentate between packet loss caused by congeston, and that due to wreless channel errors. There are two key observatons behnd ther approach; frst, relatve one way delay ncreases onotoncally f there s congeston, second, nter-arrval te s expected to ncrease f there s packet loss caused by wreless channel errors. However, the hgh wreless error sclassfcaton rate ay result n under-utlzng the wreless bandwdth, as shown n [7]. Yang et. al. also propose a slar approach to prove vdeo streang perforance n presence of wreless error, under the assupton that wreless lnk s the bottleneck [3]. Yang et.al. propose a cross-layer schee that uses lnk layer nforaton to deterne whether a packet loss s caused by channel error or congeston, assung that only the last lnk s wreless [4]. A slar assupton s ade by Akan and Akyldz n [5], to derve a wreless TFRC-lke equaton based protocol to facltate vdeo streang. The dsadvantage of end-to-end statstcs based approaches s that congeston detecton schees based on statstcs are not suffcently accurate, and they ether requre cross layer nforaton or odfcatons to the transport protocol stack. It s also possble to enable the routers wth ECN arkngs capablty to do rate control usng ECN as the easure of congeston [6]. As packet loss no longer corresponds to congeston, ECN based rate control does not adjust sendng rate upon observng a packet loss. MULTCP [7] and NetAnts [8], open ultple connectons to ncrease throughput, MULTCP was orgnally used to provde dfferental servce, and was later used to prove the perforance n hgh bandwdth-round-trp te product networks and NetAnts acheves hgher throughput by openng ultple connectons to copete for bandwdth aganst others, but openng ore connectons than needed n wred Networks ncreases the end-to-end packet loss rate experenced by end host and there s no echans to control the nuber of connectons n NetAnts. Multple TFRC s an end to-end rate control soluton for wreless vdeo streang [8]. The an dfferences between Multple TFRC and our approach are as follows: Frst, n our ethod, we use a new throughput equaton and the transsson rate s functon of soe other paraeters such as α and β n addton to pervous entoned paraeters, whch are able to provde flexble and sooth transsson rate and slowly responsble congeston control. Second, we use soothed sendng rate for new connectons. the sendng rate (X) s calculated as a functon of round trp te (RTT), loss event rate (p), packet sze (s) and new control paraeters (α, β) [9], whch are able to provde flexble and sooth transsson rate and slowly responsble congeston control and also adaptablty to unpredctable wreless channel condtons. RTT p( δ ) + 3RTO α ( p( δ )( p( + 3 p ) αδ In equaton (), k s a constant factor between 0.7 and.3 [] and RTO (retranssson teout) s equal to 4RTT [3], andδ = / β, so we have : RTT p( δ ) p( δ )( + p ( + 3p ) α( αδ By replacng α= and β=0.5 we have the followng equaton whch s well known TFRC splfed throughput equaton. p 3p (3) RTT + p ( + 3 p ) 3 8 Accordng to [0] the relatonshp between α and β to be TCP-frendly s: 4( β ) α = (4) 3 Ths relatonshp offers a wde selecton of possble values for α and β to acheve desred transent behavours, such as responsveness and reduced rate fluctuatons. We use the typcal scenaro for streang vdeo over UMTS s shown n Fgure. It s nclude a UMTS rado cell covered by a Node B connected to a rado network controller (RNC) and user equpent (UE) connected to a b-drectonal channel (DCH) to transt packet data and s reserved only for a sngle user. The coon channels are the forward access channel (FACH) n the downlnk and the rando access channel (RACH) n the uplnk. () () III. TFRC MECHANISM WITH NEW EQUATION FOR UMTS The equaton used n TFRC reflects the throughput behavour of a TCP flow, n ths way any equaton that odels realstcally a TCP flow could be used. In our ethod we use a novel rate estaton forula presented n (). It s based on a odfed verson of the TCP Reno throughput equaton (whch s desgned to copete farly wth TCP) and Fg..Typcal scenaro for streang vdeo over UMTS It s assued that the wreless lnk has avalable bandwdth Bw, and packet loss rate Pw, caused by wreless 534

3 channel error. So the axu throughput that could be acheved n the wreless lnk s Bw(-Pw). There could also be packet loss caused by congeston at wred nodes (GGSN, SGSN, RNC, NodeB). Ths eans that the throughput s X(-P) and P s end-to-end packet loss rate observed by the recever. Under the above assuptons, the wreless channel as underutlzed f X ( P) Bw( Pw). We use the TFRC odel descrbed n () to analyze the proble. Accordng to [], P s coputed as follows: P = P GGSN + (-P GGSN) P SGSN + (-P GGSN ) (- P SGSN) =P RNC + (-P GGSN ) (-P SGSN ) (-P RNC ) P Node B + (- P GGSN ) (- P SGSN ) (-P RNC ) (- P Node B) P w We also assue the followng: There s no congeston due to streang traffc to the nodes GGSN, SGSN, RNC and Node B. There s no queung delay caused at Node B, so the RTT has the nu value. The backward route s congeston-free. Downlnk bt rate Bw s 384 (kbps) and Uplnk bt rate s 8 (kbps). Downlnk TTI s 0(s) and Uplnk TTI s 0(s). Node to GGSN bandwdth s 0(Mbps) and average delay s 5(s). GGSN to SGSN bandwdth s 6(Mbps) and average delay s 0(s). SGSN to RNC bandwdth s 6(Mbps) and average delay s (s). RNC to NodeB bandwdth s 6(Mbps) and average delay s 5(s). RTCP report rate every s, Packet sze S s 800(bytes) and Pw s vares fro 0 to < β <, δ = / β, and accordng to TCP Frendly behavor 0 < α <. 33. The councaton between the sender and the recever s based on RTP/RTCP sessons. The sender (server), use the RTP protocol to transt the vdeo strea and the oble user (recever) n recurrent te space sends RTCP reports to the sender. These reports contan nforaton about the current condtons of the wreless lnk durng the transsson of vdeo between the server and the oble user. The server usng the feedback nforaton and estates the approprate rate of the streang vdeo to avod network congeston. Fro () and (6) the server s responsble for adjustng the sendng rate wth the calculated value and estates the soothed transsson rate. X = w. X + So t s essental to keep a hstory of the prevous calculated values for the transsson rate. We have chosen to keep track of eght values accordng to [] Thus, we use =8 and w,,,,0.8,0.6,0.4,0.. The server the values for weghts { } w (5) (6) extracts the feedback nforaton fro the RTCP report and passes t through an approprate flter. The use of flter s essental for the operaton of the echans n order to avod wrong estatons of the network condtons and for prevent a sngle spurous packet loss havng an excessve effect on the packet loss estaton, by usng (7) the server soothes the values of packet loss rate. P = w. P + We assue the delay te of server to clent s denoted by Dsc and the delay te of clent to server s denoted by Dsc, so RTT=Dsc+Dcs. The clent uses the testap of the receved RTP packet and the local te whch t s receved n order to estates the Dsc. We dont have the sae delay nto both drectons for UMTS, so Dsc Dcs and RTT Dsc, thus we need to estate the accurate RTT. By usng θ paraeter we can say RTT accurate = ( + θ ) Dsc. The sender estates the RTT every te t receves a recever report fro the saple clent by usng the followng equaton: RTT saple w = (rxte - txte) - delayte In equaton (8), rxte s te the feedback packet was receved by the sender, txte s transsson te of last receved data packet at the recever and delayte s te at recever between recepton of last packet and transsson of feedback packet, so we can estate the paraeter θ usng the followng equaton: RTT saple Dsc θ = (9) Dsc IV. STRATEGY OF USING MORE CONNECTIONS TO ACHIEVE MAXIMUM THROUGHPUT Equaton (5) shows that Pw s a lower bound for P and f the wreless bandwdth s underutlzed ( X Bw), then f we gnore the packet loss rate at the GGSN, SGSN, RNC and, NodeB caused by cross congeston, then RTT = RTT and P = Pw Mn, so fro () after we adjust α and β paraeters, an upper bound on the sendng rate of one TFRC connecton X ), can be derved as follows: X ub = RTT Mn ( ub w w pw( δ ) + p α( pw( δ )( ( + 3 p αδ It s obvous that X ub X and for one connecton throughput s equal to X ub ( Pw). If X ub Bw, then the wreless lnk s underutlzed and If we use ultple sultaneous connectons for a gven streang applcaton, the total throughput s expected to (7) (8) (0) ) 535

4 ncrease wth the nuber of connectons untl t reaches the hard lt of Bw(-Pw). Let us now consder the case wth two TFRC connectons fro Node to UE n Fgure. In ths case the aggregate throughput upper bound for both of the s X ub ( Pw), because of Pw for each of the two TFRC connectons rean unchanged. A slar analyss can be use for three or ore TFRC connectons, except that the wreless channel s no longer underutlzed, RTT RTT and P Pw. In ths case the Mn wreless lnk s stll fully utlzed and X (-P) =Bw(-Pw), but RTT s no longer at the nu value. When full wreless channel utlzaton occurs, the axu throughput n wreless lnk s equal to N onc. X ub ( Pw) and N onc s the optal nuber of connectons so we can say: β>0.8 unsutable for bt-rate tunng of ulteda streas due to ts nstablty. The lnearty between akes β sutable as a tunng paraeter. Fgure also shows the TCP-Frendly ponts that we use n our sulaton to change the sendng rate. Bw( Pw) = N. X ( Pw) onc ub () In equaton (), N onc s an nteger value and so we can use () for the optal nuber of connectons. N If we open ore than Bw = () X ub N onc connectons ( N ), and we onc denoted the round trp te by RTT and end-to-end packet loss rate by P, so fro () we have: Fg..The effect of varyng α paraeters on throughput for varous values of β RTT p ( δ ) + p α( p ( δ )( ( + 3 p αδ ) (3) So we can say: Bw( Pw) N = X ( P ) (4) Snce N > N onc, f we copare the above equaton wth (), we fnd out openng ore than N onc connectons results n larger RTT and hgher packet loss rate. V. ANALYTIC AND SIMULATION RESULTS To valdate the above analyss, we carry out experents usng MATLAB and NS- sulator []. We deterne how each of the paraeters affects the overall throughput; we show that by openng approprate nuber of connectons the throughput wll be ncreased. Fgure, evaluates the effect varyng α n the nterval has on throughput for dfferent values β 0.,0.3,0.4,0.5,0.6,0.7,0.8. of { } Next, the effect of varyng the β between for α 0.,0.4,0.6,0.8,,.,.33 s evaluated. dscrete values of { } The result of ths analyss s llustrated n Fgure 3 shows that the throughput grows exponentally when 0<β<. Infnte throughput s experenced as β converges on. Ths akes Fg. 3.The effect of varyng β paraeters on throughput for varous values of α We have sulated for 00 seconds and the server ntally transts the vdeo wth 56 kbps. We ntally use α= and β=0.5 to adjust the transsson rate of the vdeo. When the overall sendng rate s ncreased, we observe ncreased packet losses due to congeston, so we easured ths packet loss rate accordng to (5) and (7). Then we estate the new transsson rate accordng to () by varyng paraeters. The server estates the accurate RTT and calculates soothed transsson rate by usng the ost recent values of the calculated sendng rate accordng to (6). When the server observes an ncrease n the packet loss rate, t decreases the sendng rate of the vdeo to avod network 536

5 nstablty. The vdeo throughput n wreless lnk s showed n Fgure 4, wth the average value of 06.6 kbps. Fg. 5.The optal nuber of connectons Fg. 4.The vdeo throughput n wreless lnk In other experent we open ore connectons fro Node to UE wthout changng α and β paraeters and we could ncrease the average value of throughput by openng connecton, but Pw and RTT are ncreased, after that we open another connecton. It sees the total of the throughput s less than connectons but the packet loss rate and RTT are ncreased. The total throughput, packet loss rate and round trp te are easured (see Table ). Clearly, the optal nuber of connectons s. Specfcally, the loss rate s slghtly hgher for 3 connectons than for, whle the throughput s ore or less the sae for and 3 connectons. Table- Results for openng ore connectons Nuber of Connectons Throughput(kbps) RTT(s) P one two = three = In Fgure 5 we show the optal nuber of connectons as a functon of wreless lnk packet loss rate (Pw). We need to open ore connectons to copete for hgher Pw. In General, Ths ethod n addton to network stablty ncreases throughput wth low fluctuaton and acheves desred transent behavors. VI. CONCLUSION Rate control s an portant ssue n vdeo streang applcatons. In ths work, we focus on a echans for equaton based congeston control for vdeo transsson over UMTS networks. All prevous works on ths subject were based on the splfed TCP Reno equaton n whch sendng rate s deterned as a functon of the packet sze, the round trp te and the packet loss rate. In order to change transsson rate, packet sze or dstance between to sequental transsson ust be odfed. By changng packet sze, TFRCs characterstc wll change. In ths research another throughput equaton s used and transsson rate s functon of soe other paraeters such as α and β n addton to pervous entoned paraeters. In our ethod by usng α and β transsson rate wll be ore soothed n congested condtons and s related to loss rate. We deonstrated that hgher throughput could be acheved and t provdes soother throughput varatons. Although our ethod ncreased throughput by openng approprate nuber of connectons and ade network ore stable usng rate control, and also guarantees good qualty n delvered vdeo but we need ore syste resources for openng ore connectons. As a future work by usng large recever sde bufferng and autoatc confguraton of α and β we can prove qualty of servce. Another suggeston for future work s farness n bandwdth allocaton accordng to user requreents and consderng the stablty ssues and exanng the perforance when both the nuber of connectons and the sendng rate of each connecton are changng dynacally. REFERENCES [] Antonos G. Alexou, Dtros Antonells, Chrstos Bouras: "Adaptve and relable vdeo transsson over UMTS for enhanced perforance", Int. J. Councaton Systes 0(): 65-8 (007). 537

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