程序应该是自已explainatory如果不，用户手册是attacted。 原始代码被评论。 文件附有了： phonev2a.c...

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

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

一段简单代码123131 312 3123-a simple code 123131 312 312 3123

分网络传输部分、音频输入、音频输出、视频输入、视频输出、音频视频压缩、音频波形显示、界面。共创建个线程，音频输入线程、音频输出线程（这两个有WINAPI隐式创建）、音频视频压缩传送线程和主线程。因为音频的采样率比视频采样率高得多，这里的音频的采样率为8000Hz而视频仅为20左右，所以从宏观来看，音频是连续的，而视频是非连续的，这也就是无须为视频输入输出创建独立线程的原因。- Minute network transmission part, audio frequency input, audio frequency output, video input, video output, audio frequency video frequency compression, audio frequency profile demonstration, contact surface. Altogether founds the line regulation, the audio frequency input line regulation, the audio frequency output line regulation (these two has WINAPI to hide type to found), the audio frequency video frequency compression transmission line regulation and the master line regulation. Because the audio frequency sampling rate is much higher than the video frequency sampling rate, here audio frequency sampling rate is 8000Hz but the video frequency is only about 20, therefore from macroscopic looked, the audio frequency is continual, but video frequency right and wrong continual, this also

KPhone is a SIP (Session Initiation Protocol) user agent for Linux, with which you can initiate VoIP (Voice over IP) connections over the Internet. KPhone requires KDE (K Desktop Environment) or at least various libraries from the KDE package. KPhone also requires the OpenSSL libraries to be present in the system. KPhone has been somewhat tested on Red Hat 7.2 and 7.3, Slackware 8.0, SuSE 8.1 and Debian. Requirements on each distribution vary according to the installation layout.

基于语音传输的电话会议系统源代码，非常值得借鉴和学习-Based on voice over the telephone conferencing system source code, very much worth learning and learning

Objective Systems Open H.323 for C (ooh323c) is a simple H.323 protocol stack developed in C. The ASN.1 PER messaging code was developed using the ASN1C compiler using a modified version of our core run-time libraries. Additional open source components as well as code developed in-house were added to produce a functioning stack.

MiniSIP 网络电话开发库，为VOIP迅速开发VOIP应用提供第三分开发。 为了便于开发者方便地开发出具有各种个性化界面的软电话，本DLL COM控件不提供主用户界面，而是将主用户界面完全交付开发者根据各自的需求实现。开发者可以利用本DLL COM容易地实现各种形式的SIP软电话，例如，可以实现独立的可执行应用程序（.exe）、在Web页面中嵌入SIP软电话或者在诸如辅助办公系统、CRM系统，甚至网络游戏中实现一个嵌入的SIP软电话，从而为这些系统提供语音服务功能，使这些系统的功能和竞争力得以拓展和加强。由于采用了标准的语音编解码格式，基于本ActiveX开发的软电话除了可以实现软电话之间通话以外，还可以无缝地与其他VoIP网关实现通话，并可以通过中继网关与传统的PSTN实现通话，实现VoIP话务的落地，从而帮助开发者迅速开发出有特色的网络电话业务如Click-to-Talk、IP-Centrex或IP-PBX上的一号通以及其它软电话应用等。 提供高层封装的API接口，使得开发人员无需熟悉VoIP的具体技术，只要具备DLL COM应用技术即可。 包括了详细的SDK开发文档和应用代码。

a SIP proxy server and service provider

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