Video conferencing code, C/C++ Source, communication, image transmission categor...

IP电话源程序，在Unix环境下编译通过，基于h.323协议。-source IP telephony, in the Unix environment compiler, based on the H.323 agreement.

使用组播技术的音频会议 - The use group broadcasts technical the audio frequency conference

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

A8010 IP电话网守 用户手册 装帧-A8010 IP telephone network Shou bound user manual

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

linphone是运行在linux下的一个不错的软电话，希望能给大家开发带来点帮助-Linphone is running on linux a good under the soft phone, hope that I can bring to help the U.S. to develop

RTP实现代码，用C++语言实现，源代码-RTP code, using C++ language, source code

电子白板实现，java实现，欢迎指正，-Whiteboard realize, java realize, please correct me,

sip开源协议 -sip protocol sip open source open-source agreement

this sip voip source code too. i dont know why it does not work. please check and then send back to me. thanks -this is sip voip source code too. i dont know why it does not work. please check and then send back to me. thanks