基于H323标准的OPENSOURCE，是一个实现了基础协议栈的D

一个不错的实时通讯开发开源代码，可用于ip电话/视频会议，希望对大家有用，-A good real-time communication to develop open-source code, can be used for ip phone/video conferencing, in the hope that useful to everybody,

说明：网上的实现基于SIP协议通信的例子，来源于百度文库。已经经过测试，完全可以编译成功。使用平台：VS2010lib目录：库文件，eXosip2.lib, libeay32.libosip2.lib osipparser2.lib ssleay32.libinc目录：所用到库的头文件。dll 目录：动态库文件uac：用户代理客户端uas：用户代理服务器端

这是一个呼叫中心的源码哦,具体内容大家可以-This is a call center source Oh, everybody can be the specific content of

ffmpeg-0.4.6.tar.gz

SIP协议实现的软电话功能例子(仅供参考)-SIP protocol implementation example of the soft phone functionality (for reference only)

G.1070是IP电话应用情境下，音视频和多媒体质量评估的标准，利用它可以得到客观的质量函数-G.1070is a standard used for quality estimation

h.323协议的中文版，共三部分，可以作为多媒体业务的通信规范-the Chinese version of the h.323 agreement, including three parts, can serve as a multimedia business communications standards

主要通过对SIP协议的研究，根据这个协议的特点，设计出来一个基于SIP电话会议系统。用户可以通过WEB方式登录到预先建立好的会议室，进行电话会议。-primarily through the SIP, according to the characteristics of this agreement, designed a SIP-based telephone conferencing system. Web users can log on to advance ways to establish good meeting rooms, conference calling.

这是一篇关于sip协议介绍的文章,主要讲述了3个协议:sip,sdp,sap协议的工作原理,值得研究-This is a sip of agreement on the article, the story of three agreements : sip, sdp, sap the working principle of the agreement, worth studying

rtp relay,SIP中穿越NAT的一种方式，这是源代码，已经在red hat linux 9中调通过-rtp relay, cross SIP NAT a way, this is the source code, in red hat linux 9 emphasized through