International Journal of Applied Science and Technology

ISSN 2221-0997 (Print), 2221-1004 (Online) 10.30845/ijast

REAL TIME PERFORMANCE EVALUATION OF VOICE OVER IP CALL QUALITY UNDER VARYING NETWORK CONDITIONS
Aladdin Sleem, Olugbenga Olumuyiwa, Khaled Kamel

Abstract
Residential as well as business customers have been switching to Voice over Internet Protocol (VoIP) phone services. These new 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 router buffer size that can be tuned to improve call quality.There are several VoIP solutions, some of them are based on Peer-to-Peer (P2P) protocols and others use the Session Initiation Protocol (SIP). In this research, we analyzed the performance of VoIP solutions under different network conditions. Experiments were conducted using real networks with different design parameters such as link bandwidth, router buffers size, and the quality of calls was measured and compared for different VoIP solutions. The research concluded 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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