Extension And Optimization Of Embedded VoIP System Based On SIP And H.323

Voice serivces have been revolutioned by VoIP, which is more and more widespread. With some famous equipment manufacturers, organization for standardization and telecoms operators having joined VoIP, it is not only for internet add value service applicatin but also has developed into an important techonology, wich threatens to traditional telephony and is able to support numerous multimedia data services. There are two development trend of VoIP, which is H.323 created by ITU and SIP created by IETF. With its own advantage, each of they has occupied a certain maket, and this situation would maintain for a long time. So VoIP devices which support both SIP and H.323 are the main stream of VoIP maket.To provide VoIP serives, it needs to use many embedded VoIP end devices, which is small in size and low cost, including voice gateway, trunk gateway and IPPBX. The VoIP system mentioned in this paper is running on all these embeded devices, and it support both SIP and H.323.Extension of VoIP system means mainly function extension of software. It's descriped paticularly how to add new functions unrelated to SIP or H.323 to VoIP system and how to make VoIP system compatible with new protocol standard.Optimization of VoIP system includes overall optimization of some function module and a new advanced VoIP software system framework.This paper focuses on the new software framework, including requirement analysis, overall design and module design. A universal call controll state machine is the keystone and difficulty of the new framework. Therefore the concept of call model is introduced. Reference to the already existed popular call model, a universal call model based on SIP and H.323 is carried out. What's else, VoIP end devices adopting this universal call model can also solve the problem of interworking between SIP and H.323.