Research And Application Of Technique Base On Voice Over IP

Will you transmit the speech sound of the business quality on the IP, or transport data on the PSTN (Public Switched Telephone Network)? Rapid development of data business, growing of packet network technology, and increasing of communication channel capacity, etc, bring this problem the answer: The next generation network will be base on the IP, and it will be to consist of network architecture which are diverse, synthetic and open such as speech sound, data, multimedia etc.The principle of VoIP (Voice over Internet Protocol) is not complicated: At the sending end, sample the analogue speech sound signal, code and compress, then package and transmit it over the packet network. At receiving end the data will be decoded and resume to receive the analogy signal by digital-to-analogy convert. The basic characteristics of the current data network are point-to-point, connectless, doing one's endeavor, no quality of service, etc. These characteristics do not meet the requirement of real-time services, therefore, the realization of VoIP need support of the some key technology. These technologies includes: speech sound coding and data compression, real-time transmission and control, mute compression and multicast, acoustic-echo cancellation and comfort noise generator, dynamic monitor and guarantee of quality of network service, as well as, the compatible of different network and different protocol with each other.This article mainly discusses the basic theories and related protocol and technology of speech sound communication based on IP. The key discuss is the support of real-time services that is provided by the current IP network, codec and the acoustic-echo cancellation, as well as the dynamic monitor of network communication quality of service. It gains some conclusion by compare the merit and shortcoming between the H.323 protocol and SIP protocol and analyzing their bases: RTP/ RTCP protocol. Using the JAVA language to program to implement the RTP communication of point-to-point and point-to-multipoint. It analyze the actual data packetthat be monitored and captured in real communication. Finally,to the currently control method of the quality of service,put forward a auto-adaptive control strategy of one kind of VoIP quality of service.