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Sicsophone: A low-delay Internet telephony tool
KTH, Superseded Departments, Microelectronics and Information Technology, IMIT.
KTH, Superseded Departments, Microelectronics and Information Technology, IMIT.
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.

Place, publisher, year, edition, pages
2003. 189-195 p.
Series
EUROMICRO CONFERENCE - PROCEEDINGS, ISSN 1089-6503
Keyword [en]
packet voice, playout buffer adaption, operating systems
National Category
Telecommunications Computer Science
Identifiers
URN: urn:nbn:se:kth:diva-14295ISI: 000185703900027ISBN: 0-7695-1996-2 (print)OAI: oai:DiVA.org:kth-14295DiVA: diva2:332076
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
In thesis
1. Quality aspects of internet telephony
Open this publication in new window or tab >>Quality aspects of internet telephony
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:nbn:se:kth:diva-10572 (URN)978-91-7415-313-2 (ISBN)
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

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  • apa
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