晕，VoIP用来穿透NAT的一种方式，通过三个基本的NAT

This program was designed as a GUI client for the Asterisk PBX with Digium Zaptel cards and SIP VOIP hard or softphones as extensions, it could be adapted to other functions, but It was designed for Zap/SIP users. The program will run on X and Win32. -This program was designed as a GUI client for the Asterisk PBX with Digium Zaptel cards and SIP VOIP hard or softphones as extensions, it could be adapted to other functions, but It was designed for Zap/SIP users. The program will run on X and Win32.

H323到PSTN的网关 -Gateway from H323 to PSTN

openh323开发文档-openh323 Development Documents

IP电话源程序，在Unix环境下编译通过，基于h.323协议。-source IP telephony, in the Unix environment compiler, based on the H.323 agreement.

基于 SIP 的 VOIP 电话。它包含拨号面板，Api 对于程序员来说，基于 c 语言的核心源代码和测试软件。下面列出的来源代码结构： 建立： 生成文件和 etc.console： 基于命令行的拨号器 consolecoreapi： 核心 C.支持 SIP IP 电话 protocolgtk 源代码： 主程序发射器和拨号程序援助 codepixmaps： 所有的图标和拨号程序背景图片存储的 heretester： 测试源代码进行测试此 SIP 电话

ITU_g723参考代码-ITU_g723 reference code

SIP协议栈开发（siptiger）

gnugk-2.2.2 c实现的GateKeeper 实现IP电话网守功能。-GNUgk- 2.2.2 c achieve GateKeeper IP telephone gatekeeper function.

国外牛人的毕业论文，英文版本，实时语音传输相关，其中的两个库文件，jrtplib与jvoiplib已经上传。enjoy -foreign elite s graduate paper in english.related to real time voice transimission.it has two library,jrtplib,jvoiplib.enjoy

这是我们教研室的是师兄课题，已经通过了验证，并且有产品实践过了这是控制端的-This is the Teaching and Research Section of older subjects had been passed validation and products over the practice which is controlled side of the