IP电视电话会议的实现

利用Jainsip库开发的SIP客户端样例，适合初学者使用-Jainsip database development using SIP client sample, suitable for beginners to use

一个即时聊天软件的源代码，含服务器端和客户端。-A real-time chat software source code, including server-side and client.

学习SIP(Session Initiation Protocol)协议的最佳文档。-learning SIP (Session Initiation Protocol) the best documentation.

独立软件.Sip源代码。SIP.NET是.NET Framework和Mono的会话启动协议API。API是用100%托管C代码编写的。

Portable Windows Libary

封装了Jainsip库，可用于开发SIP客户端和服务器-Encapsulates Jainsip library, can be used to develop SIP clients and servers

srtp,一个RTP开源库，内含已经用VC7编译好的工程文件.-srtp, an open source RTP library contains has been compiled with VC7 project file.

ip电话系统设计方案，里面有多个文档，韩盖面广，管理人员可供参考-ip telephone system design, there are a number of documents, covered a wide range Korea, managers for reference

语音通信控件V3.6 集成G729A压缩算法，实时传输协议，支持一对一(点对点)，一对多（广播式），多对多（会议式）的实时通话，话音清晰流畅！适用于各种常见开发平台。-V3.6 integrated voice communication control G729A compression algorithm, real-time transport protocol to support one-on-one (point-to-point), one-to-many (broadcast), many-to-many (conference style) real-time calls, voice clear and smooth! Applicable to a variety of common development platform.

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