Abstract
The Internet has evolved into a worldwide communication infrastructure and it now provides various services including Voice over IP (VoIP) or Internet Telephony [7]. VoIP involves transmission of voice packets across the IP network known as IP telephony. Internet Protocol (IP) Telephony has many issues that have to be overcome before it can be considered a rival to the existing telephony infrastructure. One such issue is the Quality of service (QoS). The use of play-out buffering at the receiver helps to improve the quality of Voice over IP (VoIP). There exists a buffering algorithm, which uses a dynamic adaptive approach. In this algorithm the playout times of voice packets are calculated using adaptive estimation of the network delays. In contrast to previous solutions, weighting factor that controls the estimation process is dynamically adjusted according to the observed delay variations. This results in higher quality estimates of network delay. The contribution of this paper is to analyze, implement and incorporate one such adaptive buffering algorithm into the Session Initiation Protocol (SIP) through which one can achieve better delay/loss trade-off and thus better call quality.
| Original language | English |
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| Title of host publication | Third IASTED International Conference on Communications and Computer Networks, CCN 2005 |
| Editors | M.Y. Sanadidi |
| Pages | 250-255 |
| Number of pages | 6 |
| Publication status | Published - 01-12-2005 |
| Externally published | Yes |
| Event | Proceedings of the Third IASTED International Conference on Communications and Computer Networks, CCN 2005 - Marina del Rey, CA, United States Duration: 24-10-2005 → 26-10-2005 |
Conference
| Conference | Proceedings of the Third IASTED International Conference on Communications and Computer Networks, CCN 2005 |
|---|---|
| Country/Territory | United States |
| City | Marina del Rey, CA |
| Period | 24-10-05 → 26-10-05 |
All Science Journal Classification (ASJC) codes
- Engineering(all)