基于语音传输的电话会议系统源代码，非常值得借鉴和学习

G726局域网语音通话程序和源代码 这是使用G726语音压缩（16kbps）和RTP进行传输的程序，因为我没有带WIFI的PPC，所以每个程序都是单独测试的，PC端和PPC端分别都工作正常。 G726编解码算法来自OpenH323.传输使用的RTP可以在RTP程序中找到讲解，这个程序主要是G726的函数。将整个 G726封装为g726_Encode和g726_Decode两个函数，参数为压缩和解压数据存储的地址指针，可以将960字节压缩到120字节和将 120字节解压为960字节。这里G726使用的时候，音频设置为8kHz，16位量化，单声道。 使用方法很简单，只用两端各自输入对方的IP，然后按下“开始对话”，就可以进行语音通信了。 PPC端的运行比较稳定，已经进行了自收自发近一个小时连续工作的测试，非常稳定，话音清晰。-G726 LAN voice calls procedures and the source code is the use of voice compression G726 (16kbps ) and RTP for the transmission process, because I did not bring WIFI the PPC, there are separate procedures for each test, PC- and PPC-were all normal work. G726 codec algorithm from the OpenH323. RTP transmission can be used in the process to find RTP stresses Xie, this procedure is mainly a function of G726. Packaging whole G726 for g726_Encode and g726_Decode two functions, Parameters for the compression and decompression of data storage address pointer, 960 bytes can be compressed to 120 to 120 bytes and bytes to

中文版SIP协议。包括RFC2327(sdp)，RFC3263(sip)，RFC3621(sip)，RFC3372(sip)，其中3261协议包括注册部分是完全版-Chinese version of SIP. Including RFC2327 (sdp) and RFC3263 (sip), RFC3621 (sip), RFC3372 (SIP), including the 3261 agreement is fully registered version

openh323-v1_15_2-src-tar.gz 才找到的，与大家共享-openh323-v1_15_2-src-tar.gz only found with the U.S. share

c++的voip应用源码，很简单的一个例子-c++ the voip application source code, a very simple example

2003年最新版ITU-T H.323协议全文，为H.323IP电话系统开发工作提供最权威、最直接的指导。-2003 the latest version of the ITU-T H.323 transcript of H.323IP telephone systems development work to provide the most authoritative and most direct guidance.

介绍通过无线环境来传输语音的一本英文原版书籍，这里用到得无线协议是802.11 -introduced through wireless environment to transmit voice of an English original books, here used in the 802.11 wireless protocol is

用于Voip的MySQL数据库接口函数库，很有用处。-for the MySQL database interface functions, the very useful.

openh323 通用库 -Universal library for openh323

应用背景This section shows and describes a call flow for a successful call using the Cisco ATA and MGCP. Note :The term Cisco ATA refers to both the Cisco ATA 186 and the Cisco ATA 188, unless otherwise stated. 关键技术Figure C-1 on page C-2 illustrates a basic call flow between two Cisco ATAs through a VocalData Call Agent. Table C-1 on page C-3 describes the action the Cisco ATA takes for each step in the call flow illustration and includes the log created by each step.

vc编写的sip代理服务器,无需其他的库，直接编译-vc prepared sip proxy server, no additional library is directly compiled