Evaluation of SIP-based VOIP in heterogeneous networks

Session Initiation Protocol (SIP) provides advanced signaling and control functionality for a wide variety of multimedia services (audio and video). This paper investigates the effect of end-to-end delay, packet loss and jitter on SIP-based VOIP calls. The investigation is carried out on various media types (10 base-T, 100 base-TX, WLAN) with various SIP registrar (inbound and outbound). To satisfy QoS requirements of VOIP, end-to-end delay should be about 50–100ms. End-to-end delay larger than 300ms is unacceptable to most callers. We propose three different scenarios to study this situation for each type of SIP registrar (inbound and outbound). Experimental results are carried out using about eleven different (SIP-based) IP telephony calls that were carried out on each of the above mentioned scenarios. In each scenario the effect of the media type on delay, packet loss and jitter are considered and also the effect of the registrar location on the overall delay and jitter are investigated, Cross correlation and auto correlation were used to evaluate the delay also The quality of the transmitted speech is subjectively tested by a number of listeners judgments which we call the voice mean opinion score (MOS)