多媒体会议以其快捷、方便、高效、低成本的特点为广大企业节约了大量时间和经济成本，已经成为企事业单位信息化建设的重要内容，对多媒体业务良好的支持和应用是下一代网络的最大特点。 SIP协议在充分借鉴了超文本传输协议(HTTP)、简单邮件传送协议(SMTP)这两个互联网上最成功的应用层协议，继承了互联网协议简单、开放、灵活等特点。由于SIP协议没有对网络语音会议有直接的支持，目前为止还没有一个真正能够实用的基于SIP协议的语音会议系统，因而对于实现一个基于SIP协议的语音会议系统的研究是非常有意义的。 本文以此为背景，首先对软交换概念做了相关介绍，同时介绍了当前主流语音会议系统使用最多的协议H.323协议，并将该协议与使用SIP协议来构建语音会议做了相应的区别，然后设计了基于SIP协议的语音会议系统的结构，以构建一个符合企业发展的语音网络。 整个系统由三部分来组成：会议控制部分、信令处理部分和媒体处理部分，文中详细介绍了系统中各个部分的设计与实现。会议控制部分是该系统的核心，该部分主要完成用户与服务器系统之间所有控制信令的交互与处理，本文中对该部分有详细介绍。信令处理部分主要完成对所有SIP消息的接收与发送，在接受控制层的业务控制下，完成对SIP消息的传送。媒体控制部分主要是完成链路建立之后的各种音频流的接收与发送及其混音。 该系统主要是在linux系统下开发，基于开源服务器Asterisk来实现。系统遵循SIP标准协议RFC3261及其相关草案。由于使用Asterisk作为平台来实现会议功能，因此我们完全可以在此基础上完全构建起一个符合企事业单位的一个语音平台，由于Asterisk的开放性，在未来会在企事业信息化过程中会扮演重要角色。

The multimedia conferences take its convenient, fast, efficient, low cost’s characteristic saved the massive time and the cost as the general enterprises, already became the Enterprises and institutions informationization construction the important content. The next generation network's most major characteristic is for the multimedia service good supports and the application. The SIP protocol has been referencing from Hyper Text Transfer Protocol (HTTP), simple mail transfer protocol (SMTP) these two Internet the most successful application layer protocol, has inherited the Internet protocol simply, open, the nimble characteristic.But the SIP protocol does not have a direct network support for network conferencing, up to now it did not have one to be able truly practical based on the SIP protocol voice conference system, thus regarding realizes one has the significance based on the SIP protocol voice conference 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 most protocol H.323 protocol, and with used this agreement the SIP protocol to construct the voice conference to do the corresponding difference. Then the article has designed based on the SIP protocol voice conference system's structure, constructed one to conform to the enterprise development speech network. The entire system is composed of three parts: the conference control section, the signaling processing part and the media processing part. In the article we introduced in detail the various parts of the system design and implementation. The conference control section is this system's core part, this part mainly completes between the user and the server system all control 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.