The Analysis And Research Of SIP Protocol

As the competition between Internet Service Providers (ISP) becomes more serious and the trend for network mixture starts to emerge, the idea of providing a quick, satisfying and convenient services for the customer has already been embedded in the ISP'smind. The purpose of the next generation network development is to provide a comprehensive service in the unifying network platform. Thus the Voice over Internet Protocol (VoIP) is a focus of current research. The Session Initiation Protocol (SIP), as the foundation of the modern VoIP system, is one of the kernel protocols of the Soft Switch. Network based on SIP has so many applications, such as call control, routing, authentication, authority etc. Also, ISP can use the existing VoIP network equipments to reduce the cost. So research into SIP is becoming more important.According to the logic function of the application of SIP in the multimedia conversation, the system is constituted by four kinds of communication elements: the user agent, the SIP proxy server, the redirection server and the SIP register server. This research work analysis each part of SIP system respectively, especially the SIP Protocol, the Session Description Protocol (SDP), Real-time Transport Protocol (RTP) and Real-time Transport Control Protocol (RTCP) and its processes.Under the LINUX environment, The SIP source codes, based on the C++ program, are compiled and are running successfully. The result shows a voice communication demo based on SIP protocol. Finally, this research is based on the IP network components, the main objective is to analyze, with the help of flow chart, how the SIP control protocol and a collection of other protocols such as SDP, RTP and RTCP protocols are locked together to process SIP call.