一个不错的实时通讯开发开源代码，可用于ip电话/视频会议，希望对大家有用，...

视频会议web系统-web video conference system

SIP协议栈开发（siptiger）

可以用vc编译的一个sip 客户端,使用一个基于osip,exosip,ortp和mediastreamer的高层库.有该库的使用文档-Vc compiler can use a sip client, based on the use of a osip, exosip, ortp and high-level Treasury mediastreamer. The use of the library documents

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

一个功能全面的基于H323（视频会议系统协议）的网守-a fully functional based on H323 (Video Conference System agreement) Gatekeeper

It s an introduction to the most popular opensource calling card platform.Its is fair to say that A2Billing when combined with Asterisk is is now a full featured telecom platform

Delphi写的IP电话源码-IP phone sourcecode writed by Delphi

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

SIP协议资料。收集与SIP协议相关的资料。是不错的参考资料。-information of sip.

SIP_ISUP，SIP ISUP互联互通协议部分，对于学习NGN软交换之间的协议流程很有帮助。-SIP_ISUP, SIP ISUP protocol interoperability of NGN Softswitch to study the agreement between the flow of great help.