sip对sip的体系结构进行了分析，对初学sip的有很大的帮助。

高质量，底带宽，实时的视频会议软件，使用了一个Cornell大学的新算法。-high-quality, end-bandwidth, real-time video conferencing software, the use of a Cornell University"s new algorithm.

linux的大大牛alan cox写的关于linux下视频编程的api函数介绍。linux下的视频编程必备文档-linux big cow alan cox writes about linux under the video frequency programming api function introduction. Under linux video frequency programming necessary documents

一个音频编码库，支持多种音频编码方法 可以用于VoIP-an audio coding the support of a variety of audio coding method can be used for VoIP

基于QTopia的SIP IP电话源码，在LInux下编译，支持DTMF,G711,GSM-Based on the SIP IP phone qtopia source, in the compiler under LInux to support DTMF, G711, GSM

视频会议系统,进行网络视频会议的源代码，应该很详细了，数据源可能得进行改动一下-video conferencing systems, video conferencing network for the source code should be very detailed, data source may need to engage you change

VOIP program in VC++, video, voice and other features

RTC下接收呼叫的例子（如果是VC用户，请新建一个APPLICATION 32空工程，然后，用ADD FILE的方法把所有的文件包含在工程中编译）-RTC receive calls example (If yes VC users, a new air APPLICATION 32 works, and then use the ADD FILE all the documents included in the project compiler)

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

中文osip介绍书籍 中文osip介绍书籍 中文osip介绍书籍 中文osip介绍书籍 中文osip介绍书籍-failed to translate

voip 中的 rtp的具体介绍，祯结构-voip rtp of the specific briefing, Tanigaki structure