Quality aspects of Internet telephony

Ian Marsh

Friday June 5th 2009 (14:00-17.00)
Royal Institute of Technology
KTH, D2, Lindstedtsv. 5, 3rd floor.

Opponent: Professor Henning Schulzrinne (Columbia University USA)

Grading Committee:
Dr. Roar Hagen (Gipscorp, Stockholm)
Prof. Carsten Griwodz (University of Oslo)
Doc. Christer Åhlund, (Lulea University of Technology)


Finding D2

Abstract:

Internet telephony has had a tremendous impact on how people communicate. Many now maintain contact using some form of Internet telephony. Therefore the motivation for this work has been to address the quality aspects of real-world Internet telephony for both fixed and wireless telecommunication. The focus has been on the quality aspects of voice communication, since poor quality leads often to user dissatisfaction. The scope of the work has been broad in order to address the main factors within IP-based voice communication.

The first 4 chapters of this dissertation constitute the background material. The first chapter outlines where Internet telephony is deployed today. It also motivates the topics and techniques used in this research. The second chapter provides the background on Internet telephony including signalling, speech coding and voice Internetworking. The third chapter focuses solely on quality measures for packetised voice systems and finally the fourth chapter is devoted to the history of voice research.

The appendix of this dissertation constitutes the research. It includes an examination of the access network, focusing on how calls are multiplexed in wired and wireless systems. Subsequently in the wireless case, we consider how to migrate calls from 802.11 networks to the cellular infrastructure. We consider the Internet backbone where most of our work is devoted to measurements specifically for Internet telephony. The applications of these measurements have been estimating telephony arrival processes, measuring call quality, and quantifying the trend in Internet telephony quality over several years. We also consider the end systems, since they are responsible for reconstructing a voice strean given loss and delay constraints. Finally we estimate voice quality using the ITU proposal PESQ and the packet loss process.

The main contribution of this work is a systematic examination of Internet telephony. We describe several methods to enable adaptable solutions for maintaining consistent voice quality. We have also found that relatively small technical changes can lead to substantial user quality improvements. A second contribution of this work is a suite of software tools designed to ascertain voice quality in IP networks. Some of these tools are in use within commercial systems today.

Dissertation introduction (89 pages)

Complete dissertation (230 pages)

Errata (1 page)

Presentation slides (30 slides) Extended presentation set (76 slides)

Included articles:

Paper A Dimensioning links for IP telephony

Bengt Ahlgren, Anders Gunnar (nee Andersson), Olof Hagsand, and Ian Marsh.
Proceedings of the 2nd IP-Telephony Workshop, New York, April 2001.

Summary:The number of IP telephony calls that can be admitted to access networks is addressed in this paper. Link dimensioning based on packet loss is one method for dimensioning links for high utilisation of networking resources whilst providing acceptable user quality. Using this approach we also show how to select router buffer sizes. We validate and compare our approaches using a mathematical model, a discrete event simulation, and a laboratory-based implementation.

The contribution of this work is a planning tool for use in dimensioning networks for voice traffic. We have established a relationship amongst the important parameters of a packet voice network: namely the speech coding, the link capacities, the number of users, the buffer sizes, and the acceptable loss rates.

My contribution: The original idea to perform such a study was mine. I implemented most of the testbed environment and the traffic generator. Within the project I supervised a masters student, Anders Gunnar (nee Andersson), who implemented the MMPP model in Matlab and corresponding simulation scripts in ns-2. Anders was co-supervised by Professor Ingemar Kaj at Uppsala university. We were assisted by Henrik Abrahamsson, Bengt Ahlgren, Olof Hagsand and Thiemo Voigt. I co-wrote the paper with Anders Gunnar and I presented it.

Paper B Modelling the Arrival Process for Packet Audio

Ingemar Kaj and Ian Marsh.
Quality of Service in Multiservice IP Networks, Milan, Italy, February 2003.

Summary: In this work, we model the arrival process of voice packets at a receiver. The assumption is that the original spacing has been disturbed by bulk data transfers and queuing behind packets of the same stream. The solution, based on a Markov model, models the delay variation of the speech packets. The packets are assumed to be subjected to network delays when travelling from source to destination. The waiting time in intermediary buffers is assumed to be exponentially distributed. The use of such a model allows silence suppression and packet losses to be incorporated; as they are independent of the network induced delay variation.

The contribution of this work is a model for a packet audio arrival process. A simple method to estimate packet loss based on observed interarrival times is also given, independent of whether silence suppression is used or not. The model was verified by measurement data.

My contribution: The idea was jointly conceived. My contribution was the measurement data and validation of the model data. I also wrote several tools to process the data. I co-wrote and presented the paper.

Paper C Sicsophone: A Low-delay Internet Telephony Tool

Olof Hagsand, Ian Marsh, and Kjell Hanson.
IEEE 29th Euromicro Conference, Belek, Turkey, September 2003.

Summary: All VoIP systems terminate with a receiver. It can be a PC, hand-held terminal, or phone. The terminal has an important role in the overall system performance. For the PC case, we look at how to reduce delay through a novel receiver buffering scheme. The solution uses the low-level features of audio hardware and a specialised jitter buffer playout algorithm. Using the sound card memory directly eliminates intermediate buffering. A statistical-based approach for inserting packets into the audio buffers is used in conjunction with a scheme for inhibiting unnecessary fluctuations in the system. For comparison we present the performance of the playout algorithm against idealised playout conditions. To obtain an idea of the system performance we give some mouth to ear delay measurements for selected VoIP applications. The proposed mechanism is shown to save 100's of milliseconds on the end to end path.

The contribution of this work is a considerable reduction in the delay added by VoIP end systems. Although many researchers have looked at optimising and reducing jitter buffer sizes, many do not implement their ideas in a real system. A key result of this work is Sicsophone, a fully functional VoIP application.

My contribution: I wrote the RTCP part of Sicsophone. I performed comparisons between the playout delay of Sicsophone and the optimal playout delay. I co-wrote and presented the paper.

Paper D Measuring Internet Telephony Quality: Where are we today?

Olof Hagsand, Kjell Hanson, and Ian Marsh.
Proceedings of IEEE Globecom: Global Internet, Rio De Janeiro, Brazil, December 1999.

Summary: Users of Internet telephony applications demand good quality audio playback. This quality depends on the instantaneous network conditions and the time of day. In this paper, we describe a scheme for measuring network quality and motivate the development of a new metric for VoIP, asymmetry, to include into quality reports.

In 1999 we reported on the findings of our first VoIP measurement study. As far as we are aware of, the jitter and asymmetry results were new within the VoIP community. The number of downloads of the data from a COST action web site exceeded 100.

My contribution:The idea, measurements, and paper were done by me. I wrote and presented the paper. The Sicsophone tool used to conduct the measurements was originally written by Olof Hagsand and Kjell Hanson with some modifications by me for the measurement work.

Paper E Wide Area Measurements of VoIP Quality

Ian Marsh and Fengyi Li.
Quality of Future Internet Services. Stockholm, Sweden, October 2003.

Summary: We have investigated the network characteristics of loss, delay and jitter for VoIP streams that are transmitted over diverse Internet paths. Based on over 24,000 sessions, taken from nine sites connected in a full-mesh configuration we report on the average quality that can be expected by a user. The VoIP quality was acceptable for all but one of the nine sites we investigated. We also concluded that VoIP quality had improved marginally since the previous study in 1999 (paper D).

The contribution of this work is a comprehensive report on the quality of Voice over IP in 2002. We defined the quality in terms of one-way delay, loss, and jitter. For three of the sites, we have been able to compare the quality from 1999 to find some trends in VoIP quality. More than 500 downloads of the data items have taken place since they were made available. The data has been used in several other studies papers B and F in this dissertation.

My contribution:The idea to improve on the measurements from 1999 (Paper D) was mine. I advised a masters student, Fengyi Li, to perform the measurement tasks. Further modifications of Sicsophone were done by me. I wrote a tool to process the measurement data. We jointly wrote the paper based on Fengyi Li's master thesis, I presented the paper.

Paper F Self-admission control for IP telephony using early quality estimation.

Olof Hagsand, Ignacio Mas, Ian Marsh and Gunnar Karlsson.
4th IFIP-TC6 Networking, Athens, Greece, May 2004.

Summary: The idea is to use packet loss statistics from paper E to potentially identify poor quality calls given only the initial seconds of a call. The application is a self-admission control scheme, which will continue or terminate a call depending on a quality threshold, determined by the acceptable loss rates of the speech coding used. If sessions themselves can determine whether entry into a system experiencing loss, then resources, frustrating conversations, and disturbance of the on-going sessions can be avoided.

The contribution of this work is a self admission control for IP telephony. The scheme does not require any network support or external monitoring schemes.

My contribution: My role in this work was in the initial discussions and providing the measurement data. Some filtering of the data was needed to begin the work, hence I wrote the initial version of the data parsing tool. We jointly authored the paper.

Paper G IEEE 802.11b voice quality assessment using cross-layer information

Ian Marsh, Juan Carlos Martin Severiano, Victor Yuri Diogo Nunes, and Gerald Q. Maguire Jr.
In 1st Workshop on Multimedia over Wireless, Athens, Greece, April 2006.

Summary: The typical conditions that VoIP users can encounter in 801.11 networks is covered in this paper. It is measurement based and takes a methodological approach to understanding quality variations in 802.11b networks. We started with simple point-to-point VoIP experiments to calibrate the terminals and software.

We calibrate we mean finding the default delays in sending and receiving packets. We progressed onto the 802.11 infrastructure mode using line of sight and inside measurements. Next non-line of sight experiments were conducted and finally re-conducted many of the measurements against competing traffic. Some simple, but effective, mechanisms are proposed to maintain acceptable VoIP quality using 802.11 networks. We used the Sicsophone tool amended with modules for getting information from the MAC layer (retransmissions).

The contribution of this work is a comprehensive study of 802.11b networks as far as voice is concerned. This includes the methodology we employed plus utilising cross layer techniques to obtain our desired results. Many of the lessons we learned were put to use in paper H.

My contribution: The ideas for the project were mine. Most of the work was carried out by two masters students who were working on different aspects of the problem, one on the MAC layer and IP interaction and the other on IP and application layer interaction. Gerald Q. Maguire Jr. co-supervised the students. We all co-authored the paper.

Paper H The design and implementation of a quality-based handover trigger

Ian Marsh, Björn Grönvall and Florian Hammer.
5th IFIP-TC6 Networking 2006, Coimbra, Portugal, May 2006.

Summary:In this work we looked at the conditions under which an on-going call could be migrated from a 802.11 to a cellular network without perceivable loss in quality. We performed measurements on the 802.11 network in order to make workable predictions of the call quality. We implemented our solution on a hand-held terminal and performed 100 handover test trials of our handover mechanism.

The contribution of this work is one part of a fully working system that allows calls to be migrated from a WiFi to a GSM network automatically.

My contribution: Björn Grönvall I jointly conceived the initial idea and jointly performed the base experiments on which the automatic trigger was designed. We co-implemented our solution. We also integrated our solution into software developed by Optimobile AB. Florian Hammer helped in the PESQ assessment of packet loss. Björn Grönvall and I wrote the paper and I presented it.

Paper I A Systematic Study of PESQ's Performance from a Networking Perspective

Martin Varela, Ian Marsh, and Björn Grönvall.
Proceedings of Measurement of Speech and Audio Quality in Networks, Prague, Czech Republic, May 2006.

Summary: We used network losses to provide an estimator of the perceived quality of a VoIP system. Using standardised samples ``damaged'' by network losses we could utilise PESQ to map losses to quality ratings. This was performed off-line enabling a table of losses to quality estimation to be loaded and used in real time. We replayed the degraded samples to test subjects in order to investigate PESQ's validity in our setting. We also compared the single sided measure (P.563) to our own findings.

The contribution of this work is a real-time single-sided metric for estimating speech quality. A systematic study of the behavior of PESQ as a function of losses has been presented. Also the variability of PESQ ratings under several different test conditions has been performed. The PESQ ratings were compared with subjective scores for a range of bursty losses.

My contribution: The contribution of this work is a real-time single-sided metric for estimating speech quality. A systematic study of the behavior of PESQ as a function of losses has been presented. Also the variability of PESQ ratings under several different test conditions has been performed. The PESQ ratings were compared with subjective scores for a range of bursty losses.