This is a C++ Builder development of video chat program, the main function is to...

版本V4.1.1,Linux上的开放源代码的IP电话软件SIP协议，主要用于语音通话，但它也支持发送文本的即时消息，即可以像ICQ那样地用文字聊天。 如果网络带宽足够的话，它还支持可视电话 Kphone是GPL的开放源代码的软件。-version 4.1.1, It is IP phone software of Linux open source code based on SIP protocol, it can be used to voice communication, and surpport text IM, same function as ICQ used to text chat. In broad band circumstance, it surpport video conference. Kphone is GPL open source software .

voip实现的原代码 rtp协议辅助工具类代码-voip realize the original code rtp protocol aids category code

这是一款利用GENESYS CTI 的 .NET控件开发的一个典型的软 Phone,具有很好的参考价值!-This is a use of GENESYS CTI

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

网络NC v1.0 源代码.rar-network NC v1.0 source code. Rar

GSM 06.10 13kbit/s RPE/LTP 语音压缩算法 -GSM 06.10 13kbit/s RPE/LTP voice compression algorithm

JRTP最新源码 不错的RTP实现 IETF RFC 3550.-The latest source JRTP good RTP realize IETF RFC 3550.

这是我们教研室的是师兄课题，已经通过了验证，并且有产品实践过了这是控制端的-This is the Teaching and Research Section of older subjects had been passed validation and products over the practice which is controlled side of the

This program was designed as a GUI client for the Asterisk PBX with Digium Zaptel cards and SIP VOIP hard or softphones as extensions, it could be adapted to other functions, but It was designed for Zap/SIP users. The program will run on X and Win32. -This program was designed as a GUI client for the Asterisk PBX with Digium Zaptel cards and SIP VOIP hard or softphones as extensions, it could be adapted to other functions, but It was designed for Zap/SIP users. The program will run on X and Win32.

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,