QoS Evaluation of Sender-BasedLoss-Recovery Techniques for VoIPIEEE Network November/December 2006Teck-Kuen Chua and David C. Pheanis, Arizona State University

M9756001 M9756008 M9756010 M9756023 M9756034 M9756035 M9756036

VoIPAudio

InternetIP SIP VOIPIP

VoIP Diagram

IP PCM FrameITU-Tvoice frameRTPRealtime Transport ProtocolUDP messageIP message IP messageIP De-package

Loss-Recovery MechanismSender-based packet-loss recovery(sender)QoS Receiver-based packet-loss recovery(packet-loss concealment, PLC)packet-loss

Audio Codec

Codec(1/2)ITU G.711 - 64 Kbps, alaw/ulaw ITU G.722 - 48/56/64 Kbps ITU G.723.1 - 5.3/6.3 Kbps, 30ms/frame ITU G.726 - 16/24/32/40 Kbps ITU G.728 - 16 Kbps ITU G.729 - 8 Kbps, 10ms/frame

Codec(2/2) GSM - 13 Kbps (), 20ms/frame iLBC - 15Kbps,20ms/frame: 13.3 Kbps, 30ms/frame Speex - 2.15 to 44.2 Kbps LPC10 - 2.5 Kbps DoD CELP - 4.8 Kbps

Internet Low Bitrate Codec(iLBC)narrow band speechlinear predictive coding (LPC)ITU-TG.711(pulse code modulation, PCM)(packet-loss concealment, PLC) 15Kbps,20ms/frame13.3 Kbps, 30ms/frame

ITU-T Standard G.729A Codecconjugate-structure algebraic code excited linear-prediction (CS-ACELP)bit rate linear-prediction filter G.729AITU-TG.729 8 Kbps, 10ms/frame

Loss-Recovery Techniques

Loss-Recovery TechniquesVoIP(Sender-Base)VoIP Sender-Base

Plain Delivery(Plain Delivery)(sender-base)IPPlain Deliverypaper

InterleavingInterleaving

Interleaving

Forward Error CorrectionReed-SolomonParity encodingReed-Solomon (RS)Parity encoding ()

Forward Error CorrectionReed-SolomonRSRSVoIPRSRS

Forward Error CorrectionParity encoding(Parity encoding)Exn(n-1)(n+1)

Forward Error CorrectionpFEC (piggyback FEC)npFECnpFEC

Forward Error CorrectionpFEC

ABCB XOR (A XOR B)A

Redundant Data Transmission (1/2) RDT(Audio data)

IP

20

Redundant Data Transmission (2/2) frame20frame 12frame 23frame 3 4

Duplicate Packet (1/3)Duplicate Packet

Duplicate PacketRDT IP

Duplicate Packet (2/3)Duplicate Packet92.8 kb/siLBC78.4 kb/s G.729A

Duplicate Packet (3/3)Duplicate Packet20IP package20

10 percent random loss.

RetransmissionRetransmissionReceiver

Retransmission

250

QoS sender-based loss-recovery

Plain delivery RDT RTP Duplicate RTP

real-time VOIP sender-based loss-recovery

Duplicate-packets VOIPRetransmission IPdelayretransmission retransmission

Plain technique iLBC Plain technique G.729ARDTG.729ApFEC (n=2) G.729ATwo-frame interleavingG.729A

Packet-Loss CharacteristicsRandom lostBurst lostReal-Network Loss

Random lostRandom lostRandom lostsender-based loss-recovery kk (k)

RDT:RDTplain deliveryRDTDuplicate packet

pFEC(n+1)1

Interleaving techniqueplain techniqueInterleaving technique interleavingretransmission technique

12 (plain delivery, RDT, pFEC n=2 ) pFEC n=2plain delivery techniqueRDTpFEC

plain deliveryplain deliverykkpk(1-p)2 (k burst lengthp)

RDTRDTplain delivery(k-1) pk(1-p)2 k(k-1)RDT

pFECpFECk=1n=2[(k-1+p) pk(1-p)2]/2 [pk+1(1-p)2]/2k>1n=2 [(k+(k-1)+p) ]pk(1-p)2/2

Burst lostThe RDT technique is not as effective in an environment with burst loss as it is in the random-loss scenario.

RDT compensates for the final lost packet in any burst of lost packets, so the portion of the loss that RDT eliminates for a burst loss of k consecutive packets is (1/k).

When the network becomes congested and loses multiple consecutive voice packets, pFEC is very likely to lose more than one packet among (n + 1) packets.

A large transmission delay with the duplicate-packet method can help recover the corresponding lost data.

Burst lostRetransmission

Interleaving technique still achieves its goal of dispersing a long loss into several smaller losses in a burst-loss environment

Plain deliveryInterleaving technique

Real-Network LossAccurately modeling the loss characteristics of a real IP network is a difficult task.

We collected information over a period of several months from a corporate LAN with a mix of VoIP and TCP/IP traffic, and we collected similar information by testing over the Internet.

We injected a stream of VoIP packets at intervals of 20 ms and recorded packet losses in that stream so we could model thepattern of packet losses for a VoIP conversation in a realnetwork.

Since a real network loses packets during periods of traffic congestion, a real network may have higher rates of multiple-packet loss than we see with a random-loss scenario.

Real networks exhibit long periods with little or no loss andoccasional short periods of congestion and high loss.

The periods of congestion dominate the burst-loss profile and cause a real network with an overall loss rate of only about 1 percent to match the burst-loss profile for a random-loss network with an overall loss rate that is much higher.

In effect, a random-loss network provides a close approximation for a real network during a period of network congestion.

Audio Quality

PESQ (1/2)ITU-T P.862PESQMOS ratings

54321

PESQ (2/2)PESQPSTNIPdelayjitterpacket loss

ITU-T G.107 E-model

MOS ratings with random loss

MOS ratings with burst loss(10%)

MOS ratings with burst loss(20%)

Real-Network LossMOS ratings with network loss

Real-Network LossStatistics for real-network lossesNetwork loss with average loss profile

loss-recoveryRDTpFEC n = 2RDTpFEC RDT interleaving

Summary

sender-basedloss-recoveryVoIP

RDTG.729A

Thanks for your attention