IP电视电话会议+VC源码，非常好的IP电话会议的学习代码

使用JRTPLIB库实现RTP的传输RTP是目前解决流媒体实时传输问题的最好办法，如果需要在Linux平台上进行实时流媒体编程，可以考虑使用一些开放源代码的RTP库，如LIBRTP、JRTPLIB等。JRTPLIB是一个面向对象的RTP库，它完全遵循RFC 1889设计，在很多场合下是一个非常不错的选择，下面就以JRTPLIB为例，讲述如何在Linux平台上运用RTP协议进行实时流媒体编程。-JRTPLIB use of RTP for achieving the transmission RTP is to solve real-time streaming media transmission problems the best way, if necessary on the Linux platform for real-time streaming media programming, can consider using open-source library of RTP, as LIBRTP, JRTPLIB so. JRTPLIB is an object-oriented library RTP, which fully comply with the RFC 1889 design, in many occasions is a very good choice, following on to JRTPLIB example, on how the Linux platform on the use of RTP for real-time streaming media programming.

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.

openssl-0.9.7

国际电信的G729语音编码的主要程序，主要应用在VOIP中，在VC下已经调试成功，直接可以用来使用-International Telecommunications G729 speech coding of the main program, mainly used in VOIP, in the VC debugger already successful, can be used to direct the use of

POS前台C++程序,源代码齐全,对这方面感兴趣的朋友,可以-POS front C++ Procedures, the source code complete, interested friends, you can

如何成功的运用OPENH323 来开发商业的H.323 协议栈-how to successfully make use of OPENH323 to develop commercial H.323 protocol stack

本文档可以告诉您　如何在libosip的基础上开发即时消息-this document can tell you how libosip development on the basis of instant messaging

一个ip电话实例的实现，以vc为开发环境。-an example of the realization of telephone, vc for the development environment.

GSM 06.10 13 kbit/s RPE/LTP语音压缩-GSM 06.1013 kbit/S RPE/LTP voice compression

skype api application3