流媒体协议RTP/RTCP的C语言，可用于发展。

音视频聊天演示程序。聊天双方一个作为服务器，一个作为客户机。客户 机呼叫服务器建立连接之后，即可进行音视频聊天――双方都可以看到对方的图像， 可以听到对方的声音。 要求的系统配置：USB摄像头或数码摄像机，声卡，麦克风等。-Audio and video chat demo. Both sides as a chat server as a client. Client calls the server after establishing a connection, you can conduct audio and video chat- both sides can see each other

VOIP电话源程序-source VoIP telephone

IP电视电话会议的实现.

自己动手写的sip协议栈，SIP（Session Initiation Protocal）称为会话初始化协议，是由IETF（Internet Engineering Task Force）组织于1999年提出的一个基于IP网络中实现实时通信应用的一种信令协议。-Do-It-Yourself wrote sip protocol stack, SIP (Session Initiation Protocal) known as the Session Initiation Protocol is the IETF (Internet Engineering Task Force) organization in 1999 as a network-based real-time IP communications applications a letter the agreement.

vovada公司根据sip协议开发的关于voip通话的协议栈的源代码sip-1.3.0版-vovada companies under the development agreement sip voip calls on the protocol stack source code sip-version 1.3.0

这是voip网关h.323到gateway的程序代码-This is voip gateway H.323 gateway to the code

《ｖｃ视频会议开发技术与实例》第三章代码，H.263编码&解码　和H.264编码&解码-"vc video conferencing technology and the development of examples" of the third chapter code, H.263 coding

国际上著名的开源项目Bayonne IVR,完全基于电信的语音交互式系统应用.内含源码.-internationally renowned open-source projects Bayonne IVR entirely interactive voice telecommunications systems applications. Intron source.

ameridum Chat2.30 asp.net开发的聊天脚本程序-ameridum Chat2.30

网络电话。采用GSM语音格式，2.2KB/s的数据流量，支持HTTP, Socks4/5单一代理或二者同时代理。本版本修复了以前版本的一些小问题，增加了动态调整延时，打开、关闭话筒和声音，保存和自动加载通话地址列表，代理服务器列表，使用全部或部分本机地址开机-telephone network. GSM voice format, 2.2KB/s data traffic, supports HTTP, Socks4/5 of a single agent or combination of both agents. This version fixes previous versions of some small problem, the dynamic adjustment of delay, opening, closing and microphone voice, preservation and automatically load calls address list, the proxy list, the use of all or part of the machine addresses boot