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Quality aspects of internet telephony
KTH, School of Electrical Engineering (EES), Communication Networks.
2009 (English)Doctoral thesis, comprehensive summary (Other academic)
Abstract [en]

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 aspectsof 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 workhas been broad in order to address the main factors within IP-based voicecommunication.

The first four chapters of this dissertation constitute the backgroundmaterial. The first chapter outlines where Internet telephony is deployedtoday. It also motivates the topics and techniques used in this research.The second chapter provides the background on Internet telephony includingsignalling, speech coding and voice Internetworking. The third chapterfocuses solely on quality measures for packetised voice systems and finallythe fourth chapter is devoted to the history of voice research.

The appendix of this dissertation constitutes the research contributions.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 handover calls from 802.11 networks to the cellularinfrastructure. We then consider the Internet backbone where most of ourwork 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 stream 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 a certain voice quality in IP networks. Some of these tools are in use within commercial systems today.

Place, publisher, year, edition, pages
Stockholm: KTH , 2009. , 89 p.
Series
Trita-EE, ISSN 1653-5146 ; 2009:025
Series
SICS Dissertaion Series, ISSN 1101-1335 ; 51
National Category
Telecommunications
Identifiers
URN: urn:nbn:se:kth:diva-10572ISBN: 978-91-7415-313-2 (print)OAI: oai:DiVA.org:kth-10572DiVA: diva2:219379
Public defence
2009-06-05, sal D2, Stockholm, 14:00 (English)
Opponent
Supervisors
Note
QC 20100802Available from: 2009-05-27 Created: 2009-05-27 Last updated: 2013-09-09Bibliographically approved
List of papers
1. Dimensioning Links for IP Telephony
Open this publication in new window or tab >>Dimensioning Links for IP Telephony
2001 (English)In: In Proceedings of the 2nd IP-Telephony Workshop, pages 14-24, New York, USA, April 2001., 2001, 14-24 p.Conference paper, Published paper (Refereed)
Abstract [en]

Packet loss is an important parameter for dimensioning networklinks or traffic classes carrying IP telephony traffic. We present amodel based on the Markov modulated Poisson process (MMPP) whichcalculates packet loss probabilities for a set of superpositioned voice inputsources and link properties. We do not introduce another new model tothe community, rather try and verify one of the existing models via extensivesimulation and a real world implementation. A plethora of excellentresearch on queuing theory is still in the domain of ATM researchers,hence we attempt to highlight their validity to the IP (Telephony) community.Packet level simulations show reasonable correspondence with the predictionsof the model. Our main contribution is the verification of theMMPP model with measurements in a laboratory environment. The lossrates predicted by the model are in general close to the measured lossrates and the loss rates obtained by simulation. The general conclusionis that the MMPP-based model is a tool well suited for dimensioninglinks carrying packetised voice in a system with limited buffer space.

National Category
Telecommunications
Identifiers
urn:nbn:se:kth:diva-14293 (URN)
Note
QC 20100802Available from: 2010-08-02 Created: 2010-08-02 Last updated: 2010-08-02Bibliographically approved
2. Modelling the arrival process for packet audio
Open this publication in new window or tab >>Modelling the arrival process for packet audio
2003 (English)In: QUALITY OF SERVICE IN MULTISERVICE IP NETWORKS, PROCEEDINGS, 2003, Vol. 2601, 35-49 p.Conference paper, Published paper (Refereed)
Abstract [en]

Packets in an audio stream can be distorted relative to one another during the traversal of a packet switched network. This distortion can be mainly attributed to queues in routers between the source and the destination. The queues can consist of packets either from our own flow, or from other flows. The contribution of this work is a Markov model for the time delay variation of packet audio in this scenario. Our model is extensible, and show this by including sender silence suppression and packet loss into the model. By comparing the model to wide area traffic traces we show the possibility to generate an audio arrival process similar to those created by real conditions. This is done by comparing the probability density functions of our model to the real captured data.

Series
LECTURE NOTES IN COMPUTER SCIENCE, ISSN 0302-9743
Keyword
packet delay, WIP, Markov chain, steady state
National Category
Computer Science
Identifiers
urn:nbn:se:kth:diva-14294 (URN)000182655700003 ()3-540-00604-4 (ISBN)
Note
QC 20100802Available from: 2010-08-02 Created: 2010-08-02 Last updated: 2010-08-02Bibliographically approved
3. Sicsophone: A low-delay Internet telephony tool
Open this publication in new window or tab >>Sicsophone: A low-delay Internet telephony tool
2003 (English)In: PROCEEDINGS OF THE 29TH EUROMICRO CONFERENCE - NEW WAVES IN SYSTEM ARCHITECTURE, 2003, 189-195 p.Conference paper, Published paper (Refereed)
Abstract [en]

The end to end delay is a critical factor in the perceived quality of service for Voice over IP applications. Sicsophone is a complete VoIP system that couples the low level features of audio hardware with a standard jitter buffer playout algorithm. Using, the sound card directly eliminates intermediate buffering as well as providing fine control over timers needed by a soft real-time application such as VoIP A statistical based approach for inserting packets into audio buffers is used in conjunction with a scheme for inhibiting unnecessary fluctuations in the system. We also present mouth-to-ear delay measurements for selected VoIP applications and show that several hundreds of milliseconds can be saved by using the techniques described in this paper A prototype for both UNIX and Windows platforms has been implemented, demonstrating that our system adapts to network: conditions whilst maintaining low delays.

Series
EUROMICRO CONFERENCE - PROCEEDINGS, ISSN 1089-6503
Keyword
packet voice, playout buffer adaption, operating systems
National Category
Telecommunications Computer Science
Identifiers
urn:nbn:se:kth:diva-14295 (URN)000185703900027 ()0-7695-1996-2 (ISBN)
Conference
29th EUROMICRO Conference BELEK ANTALYA, TURKEY, SEP 01-06, 2003
Note
QC 20100802Available from: 2010-08-02 Created: 2010-08-02 Last updated: 2010-08-02Bibliographically approved
4. Measuring Internet telephony quality: Where are we today?
Open this publication in new window or tab >>Measuring Internet telephony quality: Where are we today?
1999 (English)In: Conference Record / IEEE Global Telecommunications Conference: GLOBECOM'99; Rio de Janeiro, Braz; 5 December 1999 through 9 December 1999, Rio de Janeiro, Braz, 1999, Vol. 3, no Piscataway, NJ, United States, 1838-1842 p.Conference paper, Published paper (Refereed)
Abstract [en]

Users of Internet telephony applications demand good quality audio playback. This quality is largely dependent on the instantaneous network conditions. In this paper we describe a scheme for measuring network connections as well as a motivation for including a new metric when assessing quality. The tests included a wide range of geographically distributed sites and our results can give useful feedback to users, operators and developers of Voice over IP applications. The results indicate that Internet telephony is feasible on todays Internet but we should envisage some problems if the Internet continues to grow at the rate so far.

Place, publisher, year, edition, pages
Rio de Janeiro, Braz: , 1999
Keyword
Internet, Network protocols, Telecommunication traffic, Voice/data communication systems, Audio quality, Internet telephony
National Category
Computer and Information Science
Identifiers
urn:nbn:se:kth:diva-14296 (URN)
Note
QC 20100802Available from: 2010-08-02 Created: 2010-08-02 Last updated: 2010-08-02Bibliographically approved
5. IEEE 802.11b voice quality assessment using crosslayer information.
Open this publication in new window or tab >>IEEE 802.11b voice quality assessment using crosslayer information.
2006 (English)Conference paper, Published paper (Refereed)
Abstract [en]

This paper reports on the suitability of IEEE802.11b networks for carrying real-time voice traffic, considering particularly the end terminals. More specifically we looked at such networks in different operating circumstances:an outdoor environment, an office environment,and the influence of competing traffic. Additionally wehave investigated the link protocol in combination with theapplication layer. Based on over 2500 recorded sessions,it can be generally concluded that the 802.11b protocolcan support real-time voice; particularly if the link transmissionrate is immediately lowered after an unsuccessful initial transmission. However, we did find situations where the voice quality deteriorated below commonly accepted values, such as when competing with high-rate TCP traffic,when intervening obstacles blocked the transmission path,and with certain uses of the RTS/CTS mechanism.

National Category
Computer Science Telecommunications
Identifiers
urn:nbn:se:kth:diva-14297 (URN)
Conference
1st Workshop on multiMedia Applications over Wireless Networks
Note

QC 20100802

Available from: 2010-08-02 Created: 2010-08-02 Last updated: 2017-06-07Bibliographically approved
6. The design and implementation of a quality-based handover trigger
Open this publication in new window or tab >>The design and implementation of a quality-based handover trigger
2006 (English)In: Lecture Notes in Computer Science, ISSN 0302-9743, E-ISSN 1611-3349, ISSN 0302-9743, Vol. 3976, 580-591 p.Article in journal (Refereed) Published
Abstract [en]

Wireless connectivity is needed to bring IP-based telephony into serious competition with the existing cellular infrastructure. However it is well known that voice quality problems can occur when used with unlicensed spectrum technologies such as the popular IEEE 802.11 standards. The cellular infrastructure could provide alternative network access should users roam out of 802.11 coverage or if heavy traffic loads are encountered in the 802.11 cell. Therefore, our goal is to design a handover mechanism to switch ongoing calls to the cellular network when the 802.11 network cannot sustain sufficient call quality. We have investigated load and coverage scenarios and designed, implemented and evaluated the performance of an 802.11 quality-based trigger for the handover of voice calls to the cellular network. We show that our predictive solution addresses the coverage problem and evaluate it within a real setting.

Keyword
802.11-Cellular convergence, Quality prediction, VoIP
National Category
Telecommunications
Identifiers
urn:nbn:se:kth:diva-14298 (URN)10.1007/11753810_49 (DOI)000238114800049 ()2-s2.0-33745884088 (Scopus ID)
Conference
5th International IFIP-TC6 Networking Conference, Networking 2006; Coimbra; Portugal; 15 May 2006 through 19 May 2006
Note

QC 20100802

Available from: 2010-08-02 Created: 2010-08-02 Last updated: 2017-12-12Bibliographically approved
7. A systematic study of PESQ’s behavior(from a networking perspective)
Open this publication in new window or tab >>A systematic study of PESQ’s behavior(from a networking perspective)
2006 (English)In: 5th International Conference MESAQIN 2006: MEASUREMENT OF AUDIO AND VIDEO QUALITY IN NETWORKS, 5-6 June 2006, Prague, Czech Republic., 2006Conference paper, Published paper (Refereed)
Abstract [en]

In this paper we study, in a systematic way, how the behavior of PESQ estimations varies with the networkloss process. We assess the variability of the estimations with respect to the network conditions and the speech content, and also their accuracy, by comparing the estimates with subjective assessments.

National Category
Telecommunications
Identifiers
urn:nbn:se:kth:diva-14299 (URN)
Note
QC 20100802Available from: 2010-08-02 Created: 2010-08-02 Last updated: 2010-08-02Bibliographically approved

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Output format
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