非常著名的视频压缩标准:H.263,全部的源码，非常适合视频编码的学习...

经典的带内音频dtmf产生和检测的c语言源码，检测准确率相当高-classic band audio DTMF generation and detection of c-source language, detection accuracy rate is very high

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.

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.

桌面视频会议系统中要用到的，鼠标一起抓 ，对于视频会议中要求有鼠标跟踪可以参考参考-Desktop video conferencing system to use, and the mouse the same time, the demands of video conferencing can refer to the reference mouse tracking

一个H323动态库使用的例子-a dynamic example of the use of

SIP Stacks libmsip: A C++ SIP stack for Linux developed for the miniSIP project. Homepage: http://www.minisip.org/libmsip/. -SIP Stackslibmsip: A C++ SIP stack for Linux developed for the miniSIP project. Homepage: http://www.minisip.org/libmsip/.

VoIP正在arapidly不断发展的技术，可以彻底改变可能的电信业。当wirelessdata网络实施，网络电话可能被证明是有助于收敛ofexisting固定和移动电话网络与快速增长的有线andwireless数据网络。服务的能力andQuality是两个需要无线VoIP的商业部署之前beresolved最重要的问题。本文开头的数据adetailed解释网络与问候网络建模与流量分布在一起。的VoIP，WLAN和VoIP在WLAN已经解释的优势，劣势和实施基于WLAN的VoIP的问题。不同的VoIP传输系统：RTP，RTSP，SIP和H323也进行了审查。一个8节点的移动电话系统hasbeen使用NS2和实验设计已经进行测量thesystem在提供服务的质量方面通过一系列的因素，包括多个节点，包大小和客流率。如延迟，吞吐量和packetlost参数已经测量通过WLAN来评估的VoIP的服务质量。基于一个NS2代码和differentparameters如延迟，吞吐量和数据包丢失的打印输出包括以supportthis论文。

visual C#源码，摄像头捕获,视频会议，网络互通-visual c# video caoture,Video meeting

视频会议传输协议，同行人一看就知道视频会议传输协议，同行人一看就知道-Video conferencing transmission protocol, peer-person, one can tell

音视频通信，利用mpeg压缩，运用网络传输协议传输-Audio and video communications, the use of mpeg compression, the use of network transmission protocol transmission