Document Type

Thesis

Date of Award

12-2010

School/College

College of Science, Engineering, and Technology (COSET)

Degree Name

MS in Computer Science

First Advisor

Professor Khaled Kamel

Second Advisor

Assistant Professor Aladdin Sleem

Abstract

Many residential as well as business customers have been switching to Voice over Internet Protocol (VoIP) phone services. VoIP phone services are based on the transmission of voice over packet switched IP networks. VoIP customers use their internet connection not only to connect to the Internet but also to make phone calls. VoIP service providers have always faced the challenge of providing customers with good call quality even though the IP network that carries call traffic does not provide any Quality of Service (QoS) guarantees. Delays, packet loss, jitter, and out of sequence packets are some sources of poor VoIP phone call quality. There are several network design parameters, such as link bandwidth and the size of router buffers that can be tuned to improve call quality or the quality of any other real-time application such as video streaming and video conferencing. There are several VoIP solutions, some of them are based on Peer-to-Peer protocols and others use the Session Initiation Protocol (SIP). This research surveys existing VoIP solutions and compares between them. It also analyzes the performance of these solutions under different network conditions. Experiments will be conducted using networks with 1 2 parameters such as bandwidth and router buffers size then phone calls will be made using different VoIP solutions and the call qualities will be compared. the research concludes by providing a comprehensive analysis of the results of the experiments highlighting the set of network parameters that gives the best call quality for each of the VoIP solutions under investigation.

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