这是一篇关于ip电话网关的规范的文章,主要讲述了ip电话应该遵循的国家标准,值得一看...

CS多客户端聊天工具 V1.0 可以多人在线聊天。经测试可以三十人以上，更多未测。

intel的IPP包用于语音编码的实例代码-the voice packets for source code examples

sip的学习参考文档，对初学sip的用户有较高的参考价值-sip learning reference documentation for novice users sip has a high reference value

包括 IP电话，视频监控，视频会议，mp3解码-VC++ Develop

一个可以进行远程视频的程序，通过ip连接进行视频传输。-One can program the remote video, video transmission via ip connection.

使用JRTPLIB库实现RTP的传输RTP是目前解决流媒体实时传输问题的最好办法，如果需要在Linux平台上进行实时流媒体编程，可以考虑使用一些开放源代码的RTP库，如LIBRTP、JRTPLIB等。JRTPLIB是一个面向对象的RTP库，它完全遵循RFC 1889设计，在很多场合下是一个非常不错的选择，下面就以JRTPLIB为例，讲述如何在Linux平台上运用RTP协议进行实时流媒体编程。-JRTPLIB use of RTP for achieving the transmission RTP is to solve real-time streaming media transmission problems the best way, if necessary on the Linux platform for real-time streaming media programming, can consider using open-source library of RTP, as LIBRTP, JRTPLIB so. JRTPLIB is an object-oriented library RTP, which fully comply with the RFC 1889 design, in many occasions is a very good choice, following on to JRTPLIB example, on how the Linux platform on the use of RTP for real-time streaming media programming.

经典的带内音频dtmf产生和检测的c语言源码，检测准确率相当高-classic band audio DTMF generation and detection of c-source language, detection accuracy rate is very high

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

解析cisco的skinny协议，根据信令自动保存udp语音包，实现cisco的VOIP系统监控。- Analyzes cisco the skinny agreement, the root according to what is believed makes automatically to preserve the udp pronunciation package, realizes the cisco VOIP system monitoring.

openh323 source code