一个语音通讯软件。

JRTP最新源码 不错的RTP实现 IETF RFC 3550.-The latest source JRTP good RTP realize IETF RFC 3550.

GSM 06.10 13kbit/s RPE/LTP 语音压缩算法 -GSM 06.10 13kbit/s RPE/LTP voice compression algorithm

I write testap.exe to run test aodisanoiiiiic    kiiidke ksiuujdmn jason test text

通过此程序可以对所有呼入和呼出的电话进行录音实现实时监控的功能-Through this procedure for all incoming and outgoing telephone recording realize real-time monitoring function

用java编写的电子白板程序-prepared with the procedures Whiteboard

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

不错的视频聊天源代码，借鉴参考研究目前市场常用的视频软件的工作原理-Good video chat source code, reference and learn from the market commonly used video software working principle

两个文件夹均是Eclipse下的Java项目，运行Eclipse后分别导入这两个工程就可以看运行结果。如何运行AP2APTest请查看开发指南.-Are two folders of Java under the Eclipse project, Eclipse after running into these two projects, respectively, can be seen on the results. Please check AP2APTest how to run the development of guidelines.

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

点对点语音通信控件V3.6，集成了G729A压缩算法，实时传输协议，话音清晰流畅！使用简单方便！适用于VC＋＋、VB、 dot NET平台、C++Builder、Delphi、易语言及网页应用程序等常见开发平台。-Point-to-point voice communication control V3.6, integrated G729A compression algorithm, Real-time Transport Protocol, voice clear and smooth! The use of simple and convenient! Apply to VC , VB, dot NET platform, C++ Builder, Delphi, easy language and procedures, and other common web application development platform.