Research On Improving Packet Of VOIP System Based On H.323 Protocol

In recent years, with the rapid development of Internet technology and applications on the Internet from the traditional application gradually extended to the media audio and video applications, the Internet, people with telephone exchanges become more and more popular, using IP telephony system in the realization of the function calls foundation, but also can provide value-added services, such as video telephony, high-fidelity stereo and telephone conference, IP telephone conference call remote teaching applications. IP telephony (VoIP) technology is reaching maturity and affluence network environment, significant price advantage in attracting users, making VoIP technology will become the future voice communications one of the mainstream technologies.The main content in the paper is as follows:Firstly, inquired in the concept and the key technique of VOIP system, including the technique of network delivering, the technique of medium data delivering, the technique of protocol and the technique of medium coding. Inquired in the related protocol, the standard and the operation principle of the VoIP system. The main protocol involved the H.323 protocol. Communication process adopted H.225protocol, the controlling of audio flow used H.245 protocol. The compression of the audio data used the G.711 or G.723 standard.Secondly, give the total structure diagram of the system, and the structure and the design processes of the terminal software, the briefness introduction of the Openh323 library, and give the communication process of the system. An improving VoIP interleaved packet format for ATM network approached, and this method combine RTP compressing technology. Forward Error Correction idea was added. Using the technique to lower the packet wrap which lowering the influence of voice effect.At last, adopt the network imitate software OPNET to carried on to imitate to the system, and coming to a the anticipant experiment result.This paper will be improved staggered sequence of packets in the ATM network, and use H.323-to-H.323 gateway provided by the RTP compression method to deliver VoIP packets, a number of packets to reduce the load and reduce the use of network bandwidth. Under the present circumstances would be 3 * 3 array is the most appropriate size of the array. Improved staggered approach makes it possible to sequence and header compression technology in the ATM network for an integrated action in light of the merits of these two methods, we can reduce the load packet headers and error-correction can increase the probability of formation a more suitable VoIP network environment.