Quality Aspects of audio communication

Quality aspects of audio communication

Ian Marsh

A thesis submitted to KTH,

the Royal Institute of Technology,

in partial fulfilment of the requirements for

the Licentiate of Technology degree.

May2003

Laboratory for Communication Networks

Department of Microelectronics and Information Technology

KTH,Royal Institute of Technology

Stockholm,Sweden

TRITA-IMIT-LCN AVH03:01

ISSN1651-4106

ISRN KTH/IMIT/LCN/AVH-03/01–SE

c Ian Marsh May2003

Printed by Universitetsservice US-AB2003

Quality aspects of audio communication

Abstract

The Internet is increasingly being used to carry real-time voice traffic. Users of real-time voice services are sensitive to variable audio quality.The quality of packet audio is largely determined by the mouth-to-ear delay and the packet loss.The contribution of this thesis is to provide techniques to improve the packet audio quality:dimensioning links specifically for packet voice communication,modelling the packet audio arrival process at a re-ceiver,measuring connectivity quality in wide area networks,and reducing delays in end systems.

Thefirst study investigates how to allocate capacity to voice traffic in a purely packet switched network.We study an idealised case for VoIP ses-sions,where the voice traffic is separated from the data traffic.A Markov modulated Poisson process model simulates the superposition of VoIPflows into afinite buffer.The model corresponds well with both packet level simu-lations and laboratory experiments.A second study looks at the interaction between voice and data traffic.We address the issue of how a constant rate VoIP stream is affected when multiplexed together with data traffic in router queues.We derive a Markov model which captures the effect of the random delays experienced by packet audio data,plus the affect of silence suppression at the sender and packet loss in the network.

Measurements made in1999and2002show that VoIP communication is feasible between academic sites in Europe and the United States.However, we show that network connectivity on a global scale still does not provide sufficient quality for satisfactory real-time voice communication.The data collated as part of this study is one of the largest publicly available reposi-tories of VoIP data,containing over18,000sample sessions.

The end systems also contribute to the delay of interactive voice com-munication.Absorption of the variable delay,or jitter,is necessary in a packet switched network in order to replay voice samples smoothly with-out glitches.We have shown that by moving the buffer used to absorb the jitter into the operating system,significant time savings can be achieved. We have implemented a VoIP tool,Sics o phone,which shows very low delay characteristics.

Using the above techniques,we show that hundreds of milliseconds can be saved in the delay budget of real-time voice communication,improving the audio quality considerably.The traffic models and measurement data presented in this thesis,will also enable future research into quality aspects of audio communication.

Keywords:Packetised voice,packetised audio,Voice over IP(VoIP), Quality of Service(QoS),speech quality,network measurements