SIP protocol stack development (siptiger)

pjsip 源码，易于学习pjsip 的ua调用-pjsip source, easy to learn pjsip the ua call

在ip电话或者视频通讯开发中，常常需要穿越nat，stun是个不错的选择-In the ip phone, or video communication development, often through the nat, stun is a good choice

introduce to sip h323

RealNetworks公司的Helix 的协议部分源码．-RealNetworks Helix source part of the agreement.

H.225 video-telephone protocol,specification document,for reference only

一个网络电话的例子-Examples of a network telephone

voip实现语音通话的例子，推荐大家下载！-voip voice examples realize recommend everyone to download!

IP电视电话会议的实现 -Implemention of IP videotelephone meeting

分网络传输部分、音频输入、音频输出、视频输入、视频输出、音频视频压缩、音频波形显示、界面。共创建个线程，音频输入线程、音频输出线程（这两个有WINAPI隐式创建）、音频视频压缩传送线程和主线程。因为音频的采样率比视频采样率高得多，这里的音频的采样率为8000Hz而视频仅为20左右，所以从宏观来看，音频是连续的，而视频是非连续的，这也就是无须为视频输入输出创建独立线程的原因。- Minute network transmission part, audio frequency input, audio frequency output, video input, video output, audio frequency video frequency compression, audio frequency profile demonstration, contact surface. Altogether founds the line regulation, the audio frequency input line regulation, the audio frequency output line regulation (these two has WINAPI to hide type to found), the audio frequency video frequency compression transmission line regulation and the master line regulation. Because the audio frequency sampling rate is much higher than the video frequency sampling rate, here audio frequency sampling rate is 8000Hz but the video frequency is only about 20, therefore from macroscopic looked, the audio frequency is continual, but video frequency right and wrong continual, this also

这是一个视频会议系统中发送端有关mpeg-4播放技术的一个源代码-This is a video conferencing system transmitter on MPEG-4 broadcast of a source code