应用源代码

利用JMF开发的语音聊天客户端，采用UDP传输-JMF developed using voice chat client, using UDP transmission

this sip voip source code too. i dont know why it does not work. please check and then send back to me. thanks -this is sip voip source code too. i dont know why it does not work. please check and then send back to me. thanks

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.

symbian平台下的基于SIP的简单游戏，可以方便地扩展成VoIP通话。-symbian platform simple SIP-based games, can be easily extended into a VoIP call.

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.

RealNetworks公司的HELIX的audio.部分-RealNetworks, the company verified our audio. Some

HELIX　的filesystem.rar部分-verified our part of the filesystem.rar

ameridum Chat2.30 asp.net开发的聊天脚本程序-ameridum Chat2.30

IP电话源码供大家共享,希望也能共享到其他源码-source for IP phones share the hope that can be shared to other FOSS