H323_CallingLineIdentificationPresentation.rar

本程序通过RTC(实时传输协议)实现了一个视频会议系统.-This procedure adopted by RTC (Real-time Transport Protocol) to achieve a video conferencing system.

基于DirectShow开发的一个视频和Windows桌面远程监控程序，比较简单易懂，提供了完全的C++源代码，可执行可复用，希望对大家工作上有帮助。-based on the development of a DirectShow Video and Windows desktop remote monitoring procedures, relatively simple to understand and provides a complete C source code, the executable can reuse, and I hope to work with you to help.

一个简易的sip软电话客户端源代码,包内含有原作者的说明文档,供大家学习参考-These are the resource codes of a simple sip program.In the package, there is a readme written by the authors.

voip 中的 rtp的具体介绍，祯结构-voip rtp of the specific briefing, Tanigaki structure

RealNetworks公司的Helix 的协议部分源码．-RealNetworks Helix source part of the agreement.

a SIP proxy server and service provider

一个基于H.323的视频会议终端，基于OPEN H323协议，实现了视频和音频的通信，支持H.263。适合研究OPEN H323的开发者参考。-H.323-based video conferencing terminals, based on the OPEN H323 agreement, realize the video and audio communications to support H.263. Appropriate to study the development of OPEN H323 reference.

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

这是voip一个部分,主要是用来做的数据代理,-This is a part of voip is primarily used to do data agents,

skype api application3