Low Rate 语音压缩

H323协议中文版，做voip的必看-H323 Chinese version of the agreement, so the voip Watchable

open323的服务器程序，C++编写-open323 server procedure, C compiles

IP电话会议系统，支持多人会议，分为Clien和Server两部分，并附有UML设计.-IP telephone conference systems to support people, Client and Server divided into two parts. together with UML design.

JRTP可用于组播也可用于单播，JRTP主页上有一个关于组播的客户端和服务器的例子，还有一个单播的客户端和服务器在同一个程序的例子，本人一直想做一个单播的客户端和服务器分开的例子，在想了一两天后终于想通，做完后发上来，秉承本人一贯风格，注释详尽、附上测试结果，倒不是什么难的东西，只是牛人都不愿做这样的东西，但还是对初学者非常有用，希望得此帮助的给个好评，只此一家，别无分店。-JRTP can be used for multicast can also be used for unicast, JRTP have a home page on the multicast client and server example, there is a unicast client and server program in the same example, I have always wanted to do a single broadcast of the separate client and server example, thinking about it for a finally realized after 12 days, after finishing up, and to inherit has been my style, notes in detail, together with test results, it is not a difficult thing, but cattle were are unwilling to do this kind of thing, but still very useful for beginners, I hope to have the help of praise, only this one, there is no branch.

实现局域网内点对点的音频通讯，数据编解码使用G.729,16:1的压缩比，数据传输使用UDP协议。音频模块可在公网内经行通讯-LAN communications within the peer-to-peer audio, data, use codec G.729, 16:1 compression ratio, data transmission using the UDP protocol. Audio module can be in the public network through the communication line

这个项目是一个简单的sip-Voip软电话客户端你只要配置你的sip服务器名称和传递字和代理，然后你必须知道你的客户端的所有IP地址，并有乐趣与你的朋友打电话，你也有开放的源代码，使你想做什么样的改变。祝你好运

对STP协议的介绍，详细介绍了结构，以及如何实现-STP agreement on the introduction, detailing the structure, and how to realize

open sip souce code c 语言实现可以在linux 编译-open sip souce codec language can realize linux compile

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

活动语音检测，语音开发中的重要算法，尤其是在voip中广泛使用。该算法简单明了。-voice activity detection, voice development of the algorithm, especially in the widely used voip. The algorithm is simple and clear.