Call Switching System Based On Sip

With the fast development of computer and network technology, the traditional PSTN based on circuit switch can not well satisfy peopl's requirement of the functions of network in Information Age. As customer claimed telephone not only has the ability of communications but also can implement new services such as personal service prepared and transmit multimedia data besides voice data.Softswitch is a core technology for implementing new generation of voice comunicating and multimedia data exchanging. Technology of softswitch has incarnated the thinking of separation of control/service and carrier in phone network. As the controlling entity softswitch fulfils functions of call connection, service controlling and user data management, and the service media stream is carried by IP network. The Session Initiation Protocol (SIP) proposed by the Internet Engineering Task Force (IETF) has become one of the most important technologies in softswitch network because of its special characteristics of simple, opening, flexible and extendable.This dissertation describes the characteristic, function, message structure and application of SIP, and explains the register and call process of the SIP based telephone via an experiment. Several modules in a call server (CS) which based on SIP are detail designed. Furthermore, a sub module named Inter-Processor Handover (IPH) which is specified to perfect the call handover function in a CS is implemented.Call handover module was implemented in this paper.And it meets the requirements of system design through the analysis results of the experiment.But there are some questions and shortages existed still to solve in future.1. During the handover process, there are some repeated messages among the modules in CS. The causes about this phenomenon are so many including the problem of network and the failures of exchanging messages inter the modules. To solve this problem, two plans are put forward. The first one is to add an error cause field which indicates whether does the error come from the delay of network or do the modules have no responses in the table to display the error info. Secondly, time after time of tests would be done to find out the module which repeatedly sent the messages and check out whether did it have any bugs in design or coding.2. When handover took place again three times later, the voice at both sides of the handsets breaks off for about 5 seconds, which is far longer than 1 second as the requirement analysis. Also the user can not abide it. The broken voice maybe makes the user assume the talk was over. So the IPH module must be improved to shorten the interval time of breaking off in handover process.3. Sometimes, when handover took place again two times later, call connecting in handover process fails at a rate about 10 percent, which can not satisfy the telecom's level stability. Thus, to promote the stability of CS and terminal product in handover process is the aim of quality promotion and main direction of progress.4. To improve the SIP terminal would make it possible for SIP terminal to perfect the handover in a call process between two different CSs. The key method to solve this problem is that how can the SIP terminal register to different CSs via different IP address of the CS in a call process.