Based On The Sip Inter-network Call

With the development of technology in the network and multimedia area, the business of multimedia communication, especially the VOIP technology, has become a leading element in Internet applications. VOIP is the abbreviation of Voice communication Over IP net. It refers to not merely the IP Phone, but also a comprehensive application platform that provides various multimedia operations including video exchange.Most VOIP systems are based on SIP or H.323 call control protocols. The Session Initiation Protocol(SIP) advanced by IETF has its distinct advantages: it is very concise in the structure, easy to be extended and more suitable to access the Internet. It can control the media gate way and thus control the PC call streaming in a multi-domain network effectively. Under the circumstance that Softswitch is to become the next generation technology of network switch, SIP is considered as the kernel protocol in access layer of the Soft switch technology, and it has been successfully applied to several representative communication systems due to the interest of lots of famous telecommunication companies.For safety or some other reasons, net are made into son net. At the borderline it places NAT/FW, which is a challege to SIP call. How SIP UA traverse NAT or Firewall, and successfully connect another UA in other son net is a problem. The problem, along with some main ideas of its solution is introduced here. This article focuses on the resolution of this problem for SIP.Detailed introduction on SIP is made firstly, then, based on the knowledge of it, a solution named "SIP Gate Way Server" is brought forward. The resolution tries to resolve the problem by modifying SIP messages in application level when they traverse NAT/FW, and controlling some behaviors of the NAT/FW. Principle and realization detail of the resolution, as well as test result is rendered in this article.