Research And Implementation Of VoIP Soft Terminal Based On SIP

With the development of computer network, packet switching and voice compressing,the voice network and the data network tend to become a whole network. Thus, VoIP (Voice over IP) Network Phone appears to satisfy needs for real- time communication.Nowadays there are two research directions of VOIP: H.323 and SIP. H.323 takes the strategy of convergence and layered control, which conforms to the traditional design idea in communication fields. However, this design idea doesn't suit for the opened and distributed Internet. SIP (Session Initiation Protocol) is a IP phone signaling protocol. Proposed by IETF as a new signaling technology, SIP is conformed to Internet. Compared with H.323 protocol, SIP has the advantages of easy implementation, high flexibility, scalability and mobility. Thus, VoIP application based on SIP protocol has much more broad prospect and is the current research focus.In the first part of this thesis, we present the research background, key technology and related protocols of VoIP. In the second, we give the detail definitions, functions, components, messages and structures of SIP. The structure, feature and useage of OSIP protocol stack are given in the third part. In the next part, we propose the design solution, framework and implementmnet of SIP stack. At last, we design and implement the sub_modules of the SIP system. The test results of the UA module show that, UA is able to establish, modify and terminate sessions according to the regulations of RFC3261.