Research And Realization Of Voice Conference System Based On SIP Protocol

The multimedia conferences take its convenient, fast, efficient, low cost's characteristicsaved the massive time and the cost as the general enterprises, already became the Enterprisesand institutions informationization construction the important content. The next generationnetwork's most major characteristic is for the multimedia service good supports and theapplication.The SIP protocol has been referencing from Hyper Text Transfer Protocol (HTTP),simple mail transfer protocol (SMTP) these two Internet the most successful application layerprotocol, has inherited the Internet protocol simply, open, the nimble characteristic.Butthe SIP protocol does not have a direct network support for network conferencing,up to nowit did not have one to be able truly practical based on the SIP protocol voice conferencesystem, thus regarding realizes one has the significance based on the SIP protocol voiceconference system's research.This article take this as the background, first introduced the concept of the soft exchange,simultaneously introduced the current mainstream voice conference system used mostprotocol H.323 protocol, and with used this agreement the SIP protocol to construct the voiceconference to do the corresponding difference. Then the article has designed based on the SIPprotocol voice conference system's structure, constructed one to conform to the enterprisedevelopment speech network.The entire system is composed of three parts: the conference control section, thesignaling processing part and the media processing part. In the article we introduced in detailthe various parts of the system design and implementation. The conference control section isthis system's core part, this part mainly completes between the user and the server system allcontrol signaling with processing, in this article has the detailed introduction alternately to this part. The signaling processing part mainly completes to all SIP messages receive and the transmission, in accepts the operational control under the service control, completes to the SIP messages transmission. After the media control section is mainly completes the link establishment each kind of audio frequency class arranges the decoding, to mix the sound and the receive and the transmission.This system is mainly develops under the Linux system, based on opens source server Asterisk to realize. The system follows SIP standard protocol RFC3261 and the related draft. Because of uses Asterisk to realize the conference function as the platform, therefore we definitely may based on this completely construct one to conform to an Enterprises and institutions'voice platform, as a result of the Asterisk openness, will play the strong character in the future in business information process.