Research And Implementation Of The NAT Traversing Over VoIP System Based On SIP

With the rapid development of the computer technology and communication technology, communication services is gradually shifting from a single voice communication into covering voice, data, image and video integrated multimedia communications. In this context, next generation network (NGN) have enjoyed great development supportting for multi-service applications and network integration in recent years. SIP-based multimedia signaling protocols of the voice and video communications have been widely used. VoIP (Voice over Internet Protocol) service is one of the important services. The most advantage of VoIP is that can make use of the global IP internetworking environment,to provide more than traditional PSTN network better, cheaper services.It is also the hot research field of Internet.The widely application of VoIP increases the requirement of a large number of IP addresses. VoIP mainly uses NAT to solve the problem of address scarcity. It is followed through the matter of the NAT traversal. The VoIP application based on H.323 and SIP uses the IP address and port in the message body of signaling to route.So besides transforming the TCP/UDP layer port and IP layer address, VoIP NAT traversal needs to transform parts of the payload of IP packages.NAT traversal is the biggest drawback of VoIP application and it should be solved soon.This thesis deeply analyzes current solutions of NAT traversal on the VoIP. However, these solutions present obvious advantages and disadvantages depending on the topological conditions of the network. So we need a complete solution that is flexible enough to through most of NAT in different network topology.Then focus the STUN technology, it presents particularly how to modify the standard the protocol of STUN to realizing traversal under different types of NATs. We designed and developed the server-side program and client-side program and test on the platform existing.