Advances in computer technology such as faster processors, better data compression schemes, and cheaper audio and video devices have made it possible to integrate multimedia into the computing environment. Desktop conferencing evolved as a plausible result of this multimedia revolution. The bandwidth granted for these conferencing applications is restricted in most cases by the speed of the modem device connected to the network.
Poor performance of multimedia conferencing over the Internet can be attributed to two main factors: local and remote induced effects. Local effects are induced by bandwidth sharing between different media components, operating system limitations, or poor design. Remote effects include all Internet related problems such as unfairness, nonguaranteed quality of service, congestion, etc. Both effects are addressed in this study and some solutions are proposed. The primary goal is to maintain audio quality and prevent video from degrading audio performance.
We study characteristics of video and audio traffic sources of conferencing applications following the H.323 set of standards defined by the International Telecommunication Union (ITU). The media traffic uses the Real-time Transport Protocol (RTP) and User Datagram Protocol (UDP) as their transport vehicle over IP network protocol. Tradeoffs involved in the choice of multimedia traffic parameters are presented. Our measurements were carried out on audio and video codecs defined in G.723.1 and H.263 specifications respectively, both drafted by the ITU.
This dissertation investigates traffic multiplexing issues at the host, and the interaction of conferencing media components as they are multiplexed locally in a shared bandwidth transport medium. Lack of appropriate multiplexing algorithms can lead to one or more media components oversubscribing to the shared bandwidth and penalizing other participants. These local effects can contribute significantly to traffic delay or abuse of the network bandwidth. We propose the “bit rate adjuster” (BRA) algorithm and use it the network bandwidth. We propose the “bit rate adjuster” (BRA) algorithm and use it for regulating media flow. The algorithm compensates for video local effects induced by packet preparation or processing to allow for better audio performance. A new performance qualifier is introduced and used in the evaluation process.
Further on the remote side, we investigate reactive mechanisms used to recover media flow performance degradation caused by shared bandwidth traffic effects. We overview feedback mechanisms based on the Real-time Control Protocol (RTCP). We uncover its limitation on applications connected to the Internet through narrow bandwidth pipes. We propose an alternative approach that predicts and prevents the loss of audio packets before it occurs based on local computation of audio jitter. We also propose a mechanism that recovers audio traffic from jitter and latency effects introduced by the Internet shared medium. These approaches improve the audio performance significantly in multimedia conferencing sessions. / Graduate
| Identifer | oai:union.ndltd.org:uvic.ca/oai:dspace.library.uvic.ca:1828/8549 |
| Date | 08 September 2017 |
| Creators | ElGebaly, Hani H. |
| Contributors | Muzio, Jon C. |
| Source Sets | University of Victoria |
| Language | English, English |
| Detected Language | English |
| Type | Thesis |
| Format | application/pdf |
| Rights | Available to the World Wide Web |
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