Graduation Semester and Year

2006

Language

English

Document Type

Thesis

Degree Name

Master of Science in Computer Science

Department

Computer Science and Engineering

First Advisor

Ramesh Yerraballi

Abstract

Voice over Internet Protocol (VoIP) or transmission of real-time voice packets over the Internet is slowly emerging as a cost-effective alternative to the traditional Public Switched Telephone Network (PSTN). However, varying end-to-end delay and packet loss, which are inherent in a packet-switched network like the Internet, lead to relatively lower quality of VoIP calls. The call quality can be improved by adaptively adjusting the play-out buffer at the receiver to reduce the impact of the delay and jitter. A standard play-out strategy uses a weighted moving average of the mean and variance of network delay to adaptively set the play-out deadline. Other techniques used include adaptive adjustment of talk-spurt and silence periods. In this thesis, we simulate an adaptive play-out algorithm that uses smoothed average of network delays and analyze its performance. For the observed packet loss rate, we determine the optimum average buffering delay and compare it to the average buffering delay that the packets experience as a result of the algorithm. We then tweak the algorithm so that the average buffering delay reported by modified algorithm is nearer to the optimum value. As a result, the end-to-end delay will be reduced improving the call quality perceived by the end-user.

Disciplines

Computer Sciences | Physical Sciences and Mathematics

Comments

Degree granted by The University of Texas at Arlington

Download

Share

COinS