IP电话的VoIP设计资料. 请大家帮忙顶！

ITU g723标准参考代码-ITU standard reference source

SDP2327，中文版，学习sipHello的同学可以参考一下-SDP2327

RealNetworks公司的Helix 的客户方部分源码．-RealNetworks Helix client part of the source code.

主要通过对SIP协议的研究，根据这个协议的特点，设计出来一个基于SIP电话会议系统。用户可以通过WEB方式登录到预先建立好的会议室，进行电话会议。-primarily through the SIP, according to the characteristics of this agreement, designed a SIP-based telephone conferencing system. Web users can log on to advance ways to establish good meeting rooms, conference calling.

该程序可以用于两个人在LAN/Intranet(或者Internet)上进行视频会议。现在有许多视频会议程序，每个都有各自的性能提升技术。主要的问题是视频会议视频帧的尺寸对于传输来说太大。因此，性能依赖于对帧的编解码。 -the two procedures can be used in LAN/Intranet (or the Internet) for video conferencing. Now there are many video conference proceedings, each have their own performance enhancements. The main problem is video conferencing video frame size too large for transmission. Therefore, the performance depends on the right frame codec.

windows下的STUN服务器，已经测试通过，可以在windows下运行通过-windows under the STUN server, the test can be run through windows

h323的视频传输-h323 video transmission

OpenJSIP 使用JAIN实现了一个完整的  SIP注册服务器和SIP代理服务器，并采用分布式架构提高性能和并发数。                                                                        At the current stage of development the project includes: 请点击左侧文件开始预览 ！预览只提供20%的代码片段，完整代码需下载后查看 加载中 侵权举报

视频同步计数，实现视频传输控制信号的工业视频应用！-Video Sync count, video transmission control signals in the industrial video applications.

vic语音编解码标准，应用于voip系统-speech codec standard applied voip system