语音聊天系统采用VB开发包括客户端和服务器

sip的学习参考文档，对初学sip的用户有较高的参考价值-sip learning reference documentation for novice users sip has a high reference value

dps上面通过c裸机编程实现流媒体以太网传输-dps through c above, bare-metal programming streaming Ethernet

数据无效

一个功能全面的基于H323（视频会议系统协议）的网守-a fully functional based on H323 (Video Conference System agreement) Gatekeeper

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

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

跨平台（Linux，Windows）的开源SIP服务器端-Cross-platform (Linux, Windows) open-source SIP server-side

MPlayer-mingw32-1.0rc1.zip

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

代码来自sourceforge，在linux下的实时语音聊天程序，使用了adpcm编码，同时还使用了RTP实时传输协议，是一个很好的学习实时传输协议的程序。- The code comes from sourceforge, chats the procedure under the linux real-time pronunciation, has used the adpcm code, meanwhile has used the RTP real-time transmission agreement, is a very good study real-time transmission agreement procedure.