Abstract

Various standards organizations have considered signaling for voice and video over IP from different approaches. There are currently two standards for signaling and control of Internet telephone calls, namely ITU-T Recommendation H.323 and the IETF Session Initiation Protocol (SIP). H.323 is an umbrella standard that provides a well-defined system architecture and implementation guidelines that cover the entire call set-up, call control, and the media used in the call. SIP is a text-based protocol that was designed to work hand in hand with other core Internet protocols such as HTTP. Both protocols provide comparable functionality using different mechanisms and provide similar quality of service. While SIP is more flexible and scalable, H.323 offers better network management and interoperability. Although there are numerous industry debates about the merits of the two protocols, the truth is that both of them, along with other complementary protocols, are necessary to provide universal access and to support IP-based enhanced services. Both protocols have been widely deployed, so interworking between SIP and H.323 is essential to ensure full end-to-end connectivity. Because of the inherent differences between H.323 and SIP, accommodation must be made to allow interworking between the two protocols. In this thesis, a new system model is established for simulating and verifying interworking between SIP and H.323. Five main components of this system are modelled by SDL/MSC: H.323 endpoint, H.323 gatekeeper, SIP-H.323 interworking facility, SIP server, SIP endpoint. Two configurations have been used in this model. One is that both protocols work within the same administrative domain, the other one is that both protocols are operating in separate administrative domains. Using a series of scenarios, it has been shown that the model meets the functional specifications outlined in SIP-H323-Interworking specification documents.