Glasgow, June, 24 th -28 th 2007 Performance Evaluation of Quality of VoIP Service over UMTS-UTRAN R99 Andrea Barbaresi, Andrea Mantovani LAB Contacts: andrea.barbaresi@telecomitalia.it Via G. Reiss Romoli, 274-10148 Torino (TO), Italy
Outline Introduction VoIP within 3GPP systems VoIP bi-directional traffic model Characterization of VoIP QoS in terms of MOS Simulation results Conclusions 1
Introduction: motivations In the last few years IP based networks and broadband access have been widely deployed and are offered now at very low prices For well known reasons (e.g. higher flexibility, capex&opex reduction, fixed-mobile convergence) also the evolutionary trend of mobile systems is focusing toward all IP packet switched only transmission for any kind of service. The voice traffic, generally offered through the circuit-switched domain (PSTN) is also expected to move over the evolved all IP mobile network architectures (e.g. LTE/SAE, mobile WiMax, ) While VoIP service on the fixed network can be considered a consolidated reality with a well known behaviour, this is not true for the wireless scenario, especially in case of mobility. 2
Introduction: investigation carried out We have evaluated by means of system level dynamic simulations the impacts of the UMTS radio access network on the end-to-end QoS of VoIP The proprietary event-driven simulator RAMSET was used; it simulates all the most relevant elements involved in the provisioning of VoIP service (bidirectional traffic model, signalling protocols, UTRAN behaviour in complex network layout) UTRAN Dedicated transport CHannels (DCHs) were considered An analytical model (ref. E-model, ITU-T G.107) devoted to predict the Mean Opinion Score (MOS, ref. ITU-T P.800) was taken into account for VoIP QoS characterization This work has been partially developed within the framework of the IST- AROMA project (www.aroma-ist.upc.edu), which is partially founded by the European Community. 3
Introduction: VoIP within the heterogeneous multi-rat network considered by the IST-AROMA project AROMA project aims to devise and assess a set of specific strategies and algorithms for both the access and the core network parts that can guarantee the end-to-end QoS in the context of an all-ip heterogeneous network Advanced CRRM/RRM mechanisms leading to an optimized usage of the different RAT technologies (WLAN, GERAN, UMTS, HSDPA, HSUPA, MBMS) were proposed and analyzed GERAN BSC Core Network Iur-g BTS BSS Abis A or Iu-CS CS PSTN BTS BSC Abis MSC Gb or Iu-PS Iur-g Iu-CS RNS Iub Node-B RNC Iu-PS Iub Node-B Iur RNC UTRAN R A x IP Gb connectivity Network (e.g. GANC AP WLAN) GAN CommonElements (e.g. HSS) G i SGSN GGSN PS G o G PDF x G p G M b q CRF IMS PSTN PDN (IP networks) Other GPRS networks Heterogeneous multi-rat network architecture Results of the investigations carried out on VoIP over UTRAN R99, HSPA, WiFi reported in project s deliverables. AROMA - Advanced Resource urce Management Solutions for Future All IP Heterogeneous Mobile Radio Environments 4
VoIP within 3GPP systems: VoIP protocol stack User Plane Voice codec (AMR 12.2kbit/s @ 20ms.) RTP RTCP Control Plane SIP Adaptive Multi-Rate (AMR) chosen by 3GPP as standard codec for voice signals in 3G mobile communication systems RTP (Real-time Transport Protocol) is the transport protocol UDP UDP TCP RTCP (RTP Control Protocol) is an optional control protocol in User Plane IP IP SIP (Session Initiation Protocol) is the signalling protocol UP protocol stack imposes a total increase of 40 bytes!! (RTP header length is 8 bytes, UDP 12, IP 20 bytes) to the dimension of the AMR frame (mean length 40 bytes) 5
VoIP within 3GPP systems: service accommodation within UTRAN User Plane PDCP RLC UM MAC PHY Control Plane RRC RLC AM MAC PHY For data and signaling traffic, assumed RABs were respectively (ref. 3GPP TR25.993): 16.8/16.8 kbit/s UL/DL PS Conversational RAB specifically optimized for AMR VoIP traffic 8/8 kbit/s UL/DL PS Interactive RAB for SIP traffic In the User Plane: PDCP performs header compression of RTP/UDP/IP headers by means of ROHC algorithm RLC works in UM mode (no radio rtx) Transport channel s TTI is 20 msec. MAC payload sizes range from 80 to 336 bits to fit different codec generated frames Physical layer Power Control was set up to operate with BLER target values of 2%, 5%, 10% 6
VoIP within 3GPP systems: client architecture De-jittering buffer within VoIP client applications input stream X X X de-jitter buffer output stream AMR frames can be lost due to errors on radio blocks AMR frames available for decoding For better QoS, decoding of AMR frame during talk spurts requires decoder to have frames always available Playout interruptions (i.e. buffer underflow occurrences) may deal with a degrade on the perceived quality since they produce silence spikes (voice clips) during the conversation Investigations made with static de-jittering buffer lengths of 0, 20, 60, 180 ms. 7
Interactive bi-directional Traffic Model The conversation between two users (A and B) can be modeled as a four state Markov chain (ref. ITU-T P.59) State A: A talking, B silent State B: A silent, B talking State D: Double talking State M: Mutual silence The sojourn time in A, B, D, and M states is modeled by a random variable with exponential distribution The transition probabilities and the exponential distribution parameters are defined in ITU-T P.59 8
QoS Evaluation: MOS and E-model MOS (Mean Opinion Score) is a subjective rating method (ITU-T P.800) a group of listeners rates voice samples from 1 ( bad ) to 5 ( excellent ) E-Model is an analytical model that reflects the effects of different types of impairments on the end-to-end speech transmission performance (ITU-T G.107) each impairment factor (i.e. end-to-end delay, packet loss, etc.) can be computed separately the result of the model is the R-factor that ranges from 100 (best case) to 0 (worst case) The R factor uniquely determines the MOS 9
Simulation assumptions: UTRAN network layout NW layout consists in 36 macro cells arranged in 12 three-sectorial sites: it can be considered representatives of a general urban environment Each VoIP call is established between a generic 3G mobile user and the host inside the IP fixed network (i.e. effects of only one radio interface link was considered) Only delays within UTRAN were considered (no CN and external IP fixed network delays) The length of simulations was set up to 3000 seconds, enough to collect results with confidence intervals of 95%, which are statistically significant 10
Simulation results: delay, FER and MOS Transmission delay vs. buffer length and BLER Transmission Delay [ms] BLER = 2% BLER = 5% BLER =10% Buffer 0ms 20 20 20 Buffer 20ms 41.3 41.2 40.2 Buffer 60ms 71.3 71.3 70.3 Buffer 180ms 132.2 131.9 131.7 Frame error rate coincides with BLER since every AMR frame is carried out by a single RLC SDU and UTRAN retransmission are not used. MOS vs. buffer length and BLER 4,5 MOS vs BLER MOS Buffer 0ms Buffer 20ms Buffer 60ms Buffer 180ms 4 3,5 BLER = 2% BLER = 5% 4.04 3.35 4.03 3.32 3.99 3.28 3.91 3.21 MOS 3 2,5 2 BLER =10% 2.57 2.55 2.51 2.43 1,5 1 Fixing BLER, MOS decreases when buffer length increases should de-jittering buffer not be used? 0,5 0 2 3 4 5 6 7 8 9 10 BLER (%) 11
Simulation results: mean number of playout interruptions Playout interruptions vs. buffer and BLER Playout interruptions Buffer 0ms Buffer 20ms Buffer 60ms Buffer 180ms Playout interruptions vs buffer (values expressed relatively to the maximum). BLER = 2% BLER = 5% >3000 >3000 2688 2920 73.5 167.6 19.2 44.4 Playout blocks vs buffer length (% with respect to maximum value) BLER =10% >3000 >3000 301.3 83.5 120,00% Client s buffer is used to have AMR frames following the ones that are lost available for decoding silence spikes strongly decrease when 20 60 ms. de-jittering buffer is considered 100,00% 80,00% 60,00% 40,00% 20,00% 0,00% 100,00% ~ -50% 50,83% ~ -99% 1,18% 0,31% 0 60 120 180 buffer length (ms) 12
Simulation results: VoIP QoS in terms of MOS & playout interruptions MOS & playout interruptions vs buffer length (BLER=2%). CDF MOS values (buffer=60ms BLER=2%). MOS, playout blocks vs buffer length 1 4,5 4,4 4,3 7000 6000 0,9 0,8 0,7 4,2 5000 0,6 4,1 4 3,9 3,8 3,7 3,6 4000 3000 2000 1000 MOS Playout blocks CDF 0,5 0,4 0,3 0,2 0,1 3,5 0 0 60 120 180 buffer length (ms) 0 3,8 3,85 3,9 3,95 4 4,05 4,1 4,15 4,2 MOS VoIP QoS can be characterized by considering trade-off between MOS (that depends on delay and packet loss) and playout interruptions 13
Simulation results: call setup delay analysis (SIP over TCP) User INVITE 100 TRYING Host CDF for session setup delay Session Setup Delay Session set-up procedure Session release procedure 183 PROGRESS PRACK 200 OK UPDATE 200 OK 180 RINGING 200 OK ACK CONVERSATION BYE 200 OK Time required to setup the VoIP call may impact the QoS level CDF 1 0,9 0,8 0,7 0,6 0,5 0,4 0,3 0,2 0,1 0 3 4 5 6 7 8 9 10 11 Time [sec] Observed mean call setup delay for VoIP session was about 6.2 s. when 8 kbit/s RAB is used. Performance when varying BLER are quite constant thanks to fast radio retransmissions 14
Concluding remarks This analysis of VoIP QoS level offered by means of UTRAN dedicated transport channel shows that it is possible to have similar QoS levels than PSTN (i.e. MOS > 3.5) Characterization of VoIP QoS service and performance evaluation versus the main parameters both at application and UTRAN level permits to derive useful information to accommodate VoIP service in UMTS in the best way: BLER = 2% ( impacts on maximum cell capacity) De-jittering buffer = 60 ms. (i.e. 3 AMR frames) UTRAN DCHs soft handover fully supports mobility of VoIP users but maximum cell capacity slightly decrease with respect to CS voice (12. 2 kbit/s CS RAB ) 15
BACK UP 16
Simulation assumptions: main parameters & performance metrics Main parameters of VoIP service : Mean call duration = 120 s. Mean traffic per user = 250 merlang 12.2 kbit/s AMR codec with Voice Activity Detection (VAD) technique SIP protocol used above the TCP stack protocols IPv4 ROHC within PDCP Performance metrics (versus BLER and buffer length) MOS Transmission delay, Frame Error Rate Number of playout interruptions (i.e. number of times per session in which the de-jittering buffer becomes empty) Call setup time, call release time 17
VoIP within 3GPP systems: the AMR codec The Adaptive Multi-Rate codec chosen by 3GPP as standard codec for voice signals in 3G mobile communication systems has been considered AMR uses Algebraic Code Excited Linear Prediction (ACELP) algorithm to reduce the overall bit rate, together with other well known techniques to further improve the quality/bandwidth ratio (e.g. VAD-Voice Activity Detection, DTX-Discontinuous Transmission) AMR may operate with eight source rates: 12.2 (GSM-EFR), 10.2, 7.95, 7.40 (IS- 641), 6.70 (PDC-EFR), 5.90, 5.15 and 4.75 kbit/s 12.2 kbit/s AMR frames of about 40 bytes (or less) are generated every 20 msec. 18
The E-model and the R-factor R =, R0-Id -Ie eff + A R 0 (signal to noise ratio) represents the perceived quality in absence of impairment factors (value of 93.2 was considered for AMR codec) I d represents all impairments due to the end-to end delay I d =0.024 d+0.11 (d-177.3) u(d-177.3) where d: e2e delay, u(): Heaviside function I e,eff represents all the impairments caused by low bit rate codecs and by packet losses I e,eff = I e +(95- I e ) p/(p+bpl) Where p:packet loss rate and I e, Bpl are codec-dependent parameters (=5, =10 for AMR 12.2kbit/s) A is the Advantage Factor considered for mobile environment (=10, ref. ITU-T G.107) 19
Traffic model (parameters) Typical values taken by the traffic model (ref. Elements of interactivity in telephone conversation, Hammer, Reichl, Raake) p1 = 0.4 p2 = 0.5 p3 = 0.5 λ A λ B λ D λ M = 1/ 854ms = 1/ 854ms = 1/ 226ms = 1/ 456ms 20
SIP procedures User Host INVITE 100 TRYING 183 PROGRESS Session set-up procedure PRACK 200 OK UPDATE 200 OK 180 RINGING 200 OK ACK Session release procedure CONVERSATION BYE 200 OK 21
BLER 2% 5% 10% Transmission delay [ms] Buffer Buffer Buffer Buffer 0ms 20ms 60ms 180ms 20 41.3 71.3 132.2 20 41.2 71.3 131.9 20 40.2 70.3 131.7 Buffer 0ms 4.04 3.35 2.57 MOS Buffer Buffer 20ms 60ms 4.03 3.99 3.32 3.28 2.55 2.51 Buffer 180ms 3.91 3.21 2.43 Playout interruptions Buffer Buffer Buffer Buffer 0ms 20ms 60ms 180ms >3000 2688 73.5 19.2 >3000 2920 167.6 44.4 >3000 >3000 301.3 83.5 22
PCG PS-Domain USER Plane PCG VoIPG VoIPG PSSG VoIP Client Streaming server Voip GW TE _ APP _ TCP TE _HTTP Streaming Client RTSP/RTP TE_APP_VOIP_RTP TE_APP_VOIP_SIP TE _ APP _ UDP HOST _APP _TCP HOST _HTTP HOST APP RTS/RT HOST_APP_VOIP_ RTP HOST_APP_VOIP_ HOST_ APP _ UDP SIP TE_TCP TE_UDP HOST_TCP HOST_UDP TE_IP GGSN_IP HOST_IP UE_PDCP RNC_PDCP RNC_GTP_U SGSN _ GTP_U GGSN_GTP_U UE_RLC RNC_RLC UE_MAC RNC_MAC UE_PHY NodeB_PHY RNC_PHY UE NodeB RNC Uu Iub Iur SGSN GGSN HOST 23
PS-Domain CONTROL Plane GGSN_PDP_Context_Manager UE_SM System Information Broadcast specific protocol termination SGSN_SM SGSN_GTP_C GGSN_GTP_C UE_PMM SGSN_PMM UE_RRC NodeB_RRC RNC_RRC RNC_RANAP SGSN_RANAP UE_RLC NodeB_RLC RNC_RLC UE_MAC NodeB_MACb RNC_MAC UE_PHY NodeB_PHY frame protocol frame protocol RNC_PHY UE NodeB RNC SGSN GGSN Uu Iub NodeB signaling Iur NodeB_RRC RNC_RRC RNC_RANAP NodeB_RLC NodeB_MACb NodeB_NBAP RNC_NBAP RNC_RLC RNC_MAC NodeB_PHY RNC_PHY NodeB RNC Iub 24
RAB PS Conversational 16.8kbps UL/16.8kbps DL VoIP Higher layer RAB/Signalling RB RAB PDCP PDCP header size, bit 0 RLC Logical channel type DTCH RLC mode UM Payload sizes, bit 80, 96, 128, 136, 152, 168, 184, 200, 216, 232, 264, 280, 288, 296, 304, 328, 336 (alt 0, 80, 96, 128, 136, 152, 168, 184, 200, 216, 232, 264, 280, 288, 296, 304, 328, 336) Max data rate, bps 16800 UMD PDU header, bit 8 MAC MAC header, bit 0 MAC multiplexing N/A Layer 1 TrCH type DCH TB sizes, bit 88, 104, 136, 144, 160, 176, 192, 208, 224, 240, 272, 288, 296, 304, 312, 336, 344 (alt 0, 88, 104, 136, 144, 160, 176, 192, 208, 224, 240, 272, 288, 296, 304, 312, 336, 344) TFS TF0, bits 0x344 (alt 1x0) TF1, bits 1x88 TF2, bits 1x104 TF3, bits 1x136 TF4, bits 1x144 TF5, bits 1x160 TF6, bits 1x176 TF7, bits 1x192 TF8, bits 1x208 TF9, bits 1x224 TF10, bits 1x240 TF11, bits 1x272 TF12, bits 1x288 TF13, bits 1x296 TF14, bits 1x304 TF15, bits 1x312 TF16, bits 1x336 TF17, bits 1x344 TTI, ms 20 Coding type TC CRC, bit 16 Max number of bits/tti after channel coding 1092 Uplink: Max number of bits/radio frame before 546 rate matching RM attribute 180-220 NOTE: Alternative 1x0 is used to have CRC present in all transport formats. Header compressor should ensure that small_cid is used and that CID 0 is allocated to this RAB 25