Performance Evaluation of Quality of VoIP Service over · PDF fileCS PS CoreNetwork PDN G i...

TELECOM ITALIA GROUPICC 2007Glasgow, June, 24th-28th 2007

| ANDREA BARBARESI | TELECOM ITALIA | TELECOM ITALIA LAB

Andrea Barbaresi, Andrea Mantovani

Performance Evaluation of Quality ofVoIP Service over UMTS-UTRAN R99

Contacts:[email protected] G. Reiss Romoli, 274 - 10148 Torino (TO), Italy

1

Performance Evaluation of Quality of VoIP Service over UMTS-UTRAN R99

| ANDREA BARBARESI| TELECOM ITALIAICC 2007

TELECOM ITALIA GROUP

Outline

Introduction

VoIP within 3GPP systems

VoIP bi-directional traffic model

Characterization of VoIP QoS in terms of MOS

Simulation results

Conclusions

2




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.

3




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 (bi-directional 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.

4




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

AROMA AROMA -- AAdvanced dvanced RResesoource urce MaManagement Solutions for Future All IP Heterogeneous Mobile Radio nagement Solutions for Future All IP Heterogeneous Mobile Radio EnvironmentsEnvironments

GANC

IP connectivityNetwork(e.g.WLAN )

GANC


GERAN

RNS

RNCNode-B

BSCBTS MSCA or Iu-CS

SGSN

Gb

GGSN

CommonElements(e.g. HSS)

CS

PS

Core Network

PDN(IP networks)Gi

IMS

Gp

OtherGPRSnetworks

PSTN

Mb

PSTN

PDFGo

GqCRF

Gx

Rx

Iur-g

RNC

Iur

BSCIur-g

BTS

Node-B

Iub

Iub

Abis

Abis

BSS

UTRAN

GERAN

RNS

RNCNode-B

BSCBTS MSCor Iu-CS

Iu-CSIu-

SGSN

Gb

Iu-PSIu-PSGGSN


CS

PS

Core Network

PDN(IP networks)Gi

IMS

Gp

OtherGPRSnetworks

PSTN

Mb

PSTN

PDFGo

GqCRF

Gx

Rx

Iur-g

RNC

Iur

BSCIur-g

BTS

Node-B

Iub

Iub

Abis

Abis

BSS

UTRAN

oror Iu-PSIu-PS

Gb

A

GAN

APGANC


GANC


GERAN

RNS

RNCNode-B

BSCBTS MSCA or Iu-CS

SGSN

Gb

GGSN


CS

PS

Core Network

PDN(IP networks)Gi

IMS

Gp

OtherGPRSnetworks

PSTN

Mb

PSTN

PDFGo

GqCRF

Gx

Rx

Iur-g

RNC

Iur

BSCIur-g

BTS

Node-B

Iub

Iub

Abis

Abis

BSS

UTRAN

GERAN

RNS

RNCNode-B

BSCBTS MSCor Iu-CS

Iu-CSIu-Iu-CSIu-

SGSN

Gb

Iu-PSIu-PSIu-PSIu-PSGGSN


CS

PS

Core Network

PDN(IP networks)Gi

IMS

Gp

OtherGPRSnetworks

PSTN

Mb

PSTN

PDFGo

GqCRF

Gx

Rx

Iur-g

RNC

Iur

BSCIur-g

BTS

Node-B

Iub

Iub

Abis

Abis

BSS

UTRAN

oror Iu-PSIu-PSIu-PSIu-PS

Gb

A

GAN

AP

Results of the investigations carried out on VoIP over UTRAN R99, HSPA, WiFi reported in project’s deliverables.

Heterogeneous multi-RAT network architecture

5




VoIP within 3GPP systems: VoIP protocol stack

IP

UDP

RTP

Voice codec(AMR 12.2kbit/s

@ 20ms.)

RTCP

TCP

IP

UDP

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

RTCP (RTP Control Protocol) is an optional control protocol in User Plane

SIP (Session Initiation Protocol) is the signalling protocol

User Plane Control Plane

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)

6




VoIP within 3GPP systems: service accommodation within UTRAN

MAC

PHY

RLC UM

PDCP RRC

PHY

MAC

RLC AM

For data and signaling traffic, assumed RABswere 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

User Plane Control Plane

Physical layer Power Control was set up to operate with BLER target values of 2%, 5%, 10%

7




VoIP within 3GPP systems: client architectureDe-jittering buffer within VoIP client applications

AMR frames can be lostdue to errors on radio blocks

AMR frames available for decoding

input stream output streamde-jitter buffer

XXX

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.

8




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

9




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

10




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

11




Simulation results: delay, FER and MOS

131.770.340.220BLER =10%

131.971.341.220BLER = 5%

132.271.341.320BLER = 2%

Buffer 180msBuffer 60msBuffer 20msBuffer 0msTransmission Delay [ms]

Transmission delay vs. buffer length and BLER

2.432.512.552.57BLER =10%

3.213.283.323.35BLER = 5%

3.913.994.034.04BLER = 2%

Buffer 180msBuffer 60msBuffer 20msBuffer 0msMOS

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 BLER

0

0,5

1

1,5

2

2,5

3

3,5

4

4,5

2 3 4 5 6 7 8 9 10

BLER (%)

MO

S

MOS vs. buffer length and BLER

Fixing BLER, MOS decreases when buffer length increases should de-jittering buffer not be used ?

12




Simulation results: mean number of playout interruptions

Playout interruptions vs. buffer and BLER

83.5301.3>3000>3000BLER =10%

44.4167.62920>3000BLER = 5%

19.273.52688>3000BLER = 2%

Buffer 180msBuffer 60msBuffer 20msBuffer 0msPlayout interruptions

Playout blocks vs buffer length(% with respect to maximum value)

100,00%

50,83%

0,31%1,18%

0,00%

20,00%

40,00%

60,00%

80,00%

100,00%

120,00%

0 60 120 180

buffer length (ms)

Playout interruptions vs buffer (values expressed relatively to the maximum).

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

~ -50%

~ -99%

13




Simulation results: VoIP QoS in terms of MOS & playout interruptions

MOS & playout interruptions vs buffer length (BLER=2%).

MOS, playout blocks vs buffer length

3,5

3,6

3,7

3,8

3,9

4

4,1

4,2

4,3

4,4

4,5

0 60 120 180

buffer length (ms)

0

1000

2000

3000

4000

5000

6000

7000

MOSPlayout blocks

0

0,1

0,2

0,3

0,4

0,5

0,6

0,7

0,8

0,9

1

3,8 3,85 3,9 3,95 4 4,05 4,1 4,15 4,2

MOS

CDF

CDF MOS values (buffer=60ms BLER=2%).

VoIP QoS can be characterized by considering trade-off between MOS (that depends on delay and packet loss) and playout interruptions

14




Simulation results: call setup delay analysis (SIP over TCP)

CDF for session setup delay

Session Setup Delay

0

0,1

0,2

0,3

0,4

0,5

0,6

0,7

0,8

0,9

1

3 4 5 6 7 8 9 10 11

Time [sec]

CDF

Time required to setup the VoIP call may impact the QoS level

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

User Host

INVITE

100 TRYING

183 PROGRESS

PRACK

200 OK

UPDATE

200 OK

ACK

BYE

200 OK

180 RINGING

200 OK

CONVERSATION

Ses

sion

set

-up

proc

edur

eS

essi

on re

leas

epr

oced

ure

User Host

INVITE

100 TRYING

183 PROGRESS

PRACK

200 OK

UPDATE

200 OK

ACK

BYE

200 OK

180 RINGING

200 OK

CONVERSATION

Ses

sion

set

-up

proc

edur

eS

essi

on re

leas

epr

oced

ure

15




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 )

16




BACK UP

17




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

18




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.

19




The E-model and the R-factor

A-I-IRR effed += ,0

R0 (signal to noise ratio) represents the perceived quality in absence of impairment factors (value of 93.2 was considered for AMR codec)

Id represents all impairments due to the end-to end delayId =0.024•d+0.11•(d-177.3)•u(d-177.3)

where d: e2e delay, u(): Heaviside function

Ie,eff represents all the impairments caused by low bit rate codecs and by packet losses

Ie,eff = Ie +(95- Ie)•p/(p+Bpl)

Where p:packet loss rate and Ie , 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)

20




Traffic model (parameters)

⎪⎩

⎪⎨

⎧

===

5.035.024.01

ppp

⎪⎪⎩

⎪⎪⎨

⎧

====

msmsmsms

M

D

B

A

456/1226/1854/1854/1

λλλλ

Typical values taken by the traffic model (ref. “Elements of interactivity in telephone conversation”, Hammer, Reichl, Raake)

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SIP proceduresUser Host

INVITE

100 TRYING

183 PROGRESS

PRACK

200 OK

UPDATE

200 OK

ACK

BYE

200 OK

180 RINGING

200 OK

CONVERSATION

Ses

sion

set

-up

proc

edur

eS

essi

on re

leas

epr

oced

ure

User Host

INVITE

100 TRYING

183 PROGRESS

PRACK

200 OK

UPDATE

200 OK

ACK

BYE

200 OK

180 RINGING

200 OK

CONVERSATION

Ses

sion

set

-up

proc

edur

eS

essi

on re

leas

epr

oced

ure

22




83.5301.3>3000>30002.432.512.552.57131.770.340.22010%44.4167.62920>30003.213.283.323.35131.971.341.2205%19.273.52688>30003.913.994.034.04132.271.341.3202%

Buffer 180ms

Buffer 60ms

Buffer 20ms

Buffer 0ms

Buffer 180ms

Buffer 60ms

Buffer 20ms

Buffer 0ms

Buffer 180ms

Buffer 60ms

Buffer 20ms

Buffer 0ms

Playout interruptionsMOSTransmission delay [ms]BLER

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UE_RLC

UE_MAC

TE_IP

TE_TCP TE_UDP

NodeB_PHY

RNC_RLC

RNC_MAC

RNC_PHY

RNC_GTP_U SGSN _ GTP_U GGSN_GTP_U

HOST_IP

HOST_TCP HOST_UDP

Uu Iub

UE NodeB RNC

Iur

SGSN GGSN HOST

UE_PDCP RNC_PDCP

GGSN_IP

UE_PHY

PS-Domain USER Plane PCG

TE _ APP _ TCP TE _HTTP TE _ APP _ UDP Streaming

ClientRTSP/RTP

PCG

HOST _APP _TCPHOST _HTTP HOST_ APP _ UDP HOST APP

RTS/RT

PSSG

Streamingserver

VoIPG

TE_APP_VOIP_RTP TE_APP_VOIP_SIP

HOST_APP_VOIP_

RTP HOST_APP_VOIP_

SIP

Voip GW

VoIPG

VoIP Client

24




UE_RLC

UE_MAC

UE_PHY

UE_RRC

UE_PMM

UE_SM

NodeB_MACb

RNC_RLC

RNC_MAC

RNC_PHY

RNC_RRC RNC_RANAP SGSN_RANAP

SGSN_SM

SGSN_PMM

GGSN_GTP_C

NodeB_RRC

NodeB_RLC

Uu Iub

UE NodeB RNCIur

SGSN GGSN

SGSN_GTP_C

RNC_NBAP

NodeB_PHY

System Information Broadcastspecific protocol termination

PS-Domain CONTROL Plane

NodeB_MACb

RNC_RLC

RNC_MAC

RNC_PHY

RNC_RRC RNC_RANAP

NodeB_NBAP

NodeB_RRC

NodeB_RLC

Iub

NodeB RNC

NodeB_PHY

frame protocol frame protocol

NodeB signaling

GGSN_PDP_Context_Manager

25




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

180-220RM attribute

546Uplink: Max number of bits/radio frame before rate matching

1092Max number of bits/TTI after channel coding16CRC, bitTCCoding type20TTI, ms

1x344TF17, bits1x336TF16, bits1x312TF15, bits1x304TF14, bits1x296TF13, bits1x288TF12, bits1x272TF11, bits1x240TF10, bits1x224TF9, bits1x208TF8, bits1x192TF7, bits1x176TF6, bits1x160TF5, bits1x144TF4, bits1x136TF3, bits1x104TF2, bits1x88TF1, bits

0x344 (alt 1x0)TF0, bitsTFS

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)

TB sizes, bitDCHTrCH typeLayer 1N/AMAC multiplexing

0MAC header, bitMAC8UMD PDU header, bit

16800Max data rate, bps

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)

Payload sizes, bitUMRLC mode

DTCHLogical channel typeRLC0PDCP header size, bitPDCP

RABRAB/Signalling RBHigher layerRAB PS Conversational 16.8kbps UL/16.8kbps DL – VoIP

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