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A novel implementation of very large teleconferences

Published: 02 August 2010 Publication History

Abstract

Certain teleconferencing applications must host very large number of participants that exceeds the capacity of a single mixing media server. Traditionally multiple media servers are connected in a cascading arrangement to meet the capacity requirement. This paper discusses the shortcomings of the traditional approach such as lower audio quality and unfairness of speaker selection. It then presents a novel approach that exploits the exibility of Voice-over-IP and the Session Initiation Protocol to move participants between a main conference mixer and one or more media-distributing replicators as their role change between active talk-listen and listen-only. Benchmarking of an implementation of the replicators on general-purpose computers shows that large capacity can be achieved without specialized hardware. Moreover, it is shown how this approach can augment an already deployed teleconferencing system without modifying the existing telephony features, thereby illustrating the power of modularity and application composition in the SIP servlet environment.

References

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E. Burger and M. Dolly. A Session Initiation Protocol (SIP) event package for key press stimulus (KPML), November 2006. IETF RFC 4730.
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E. Cheung and K. Purdy. An application router for SIP servlet application composition. In Communications, 2008. ICC '08. IEEE International Conference on, pages 1802--1806, May 2008.
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R. Even and N. Ismail. Conferencing scenarios, July 2006. IETF RFC 4597.
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M. Radenkovic, C. Greenhalgh, and S. Benford. A scaleable and adaptive audio service to support large scale collaborative work and entertainment. In Proceedings of International Conference on Advances in Infrastructures for Electronic Business, Education, Science and Medicine on the Internet (SSGRR 2002), 2002.
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H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson. RTP: A transport protocol for real-time applications, July 2003. IETF RFC 3550.
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H. Schulzrinne and T. Taylor. RTP payload for DTMF digits, telephony tones, and telephony signals, December 2006. IETF RFC 4733.
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K. Singh, G. Nair, and H. Schulzrinne. Centralized conferencing using SIP. In Proceedings of the 2nd IP-Telephony Workshop (IPTel'2001), April 2001.
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    cover image ACM Other conferences
    IPTComm '10: Principles, Systems and Applications of IP Telecommunications
    August 2010
    170 pages
    ISBN:9781450306317
    DOI:10.1145/1941530
    Permission to make digital or hard copies of all or part of this work for personal or classroom use is granted without fee provided that copies are not made or distributed for profit or commercial advantage and that copies bear this notice and the full citation on the first page. Copyrights for components of this work owned by others than ACM must be honored. Abstracting with credit is permitted. To copy otherwise, or republish, to post on servers or to redistribute to lists, requires prior specific permission and/or a fee. Request permissions from [email protected]

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    Published: 02 August 2010

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    Author Tags

    1. RTP
    2. SIP servlets
    3. composition
    4. conferencing
    5. media streaming
    6. multimedia services
    7. real-time system

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    IPTComm '10
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    • Technische Universitat Munchen

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    IPTComm '10 Paper Acceptance Rate 12 of 50 submissions, 24%;
    Overall Acceptance Rate 18 of 62 submissions, 29%

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