windows/linux平台下C语言编写的sip软电话siphon最新版本。

版本V4.1.1,Linux上的开放源代码的IP电话软件SIP协议，主要用于语音通话，但它也支持发送文本的即时消息，即可以像ICQ那样地用文字聊天。 如果网络带宽足够的话，它还支持可视电话 Kphone是GPL的开放源代码的软件。-version 4.1.1, It is IP phone software of Linux open source code based on SIP protocol, it can be used to voice communication, and surpport text IM, same function as ICQ used to text chat. In broad band circumstance, it surpport video conference. Kphone is GPL open source software .

开发一套易用的网络通信包，以C＋＋形式对功能进行封装。做的相当不错，包含开发文档及代码说明 -development of a set of easy-to-use communications network packets to C++ form of a wrapper function. Doing quite well, including the development of documents and code Description

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

本文主要研究了W-WAN中GPRS网络的移动视频传输问题。文中探讨了GPRS的网络结构、协议体系以及链路特性，从中我们看出： GPRS网络链路不仅具有一般无线链路数据传输中遇到的延迟大、误码率高等共性问题，而且具有移动通信系统采用蜂窝频率复用方式带来的小区切换， 以及由于通信中链路性能随着地理位置变换而变化造成的GPRS网络中不可避免的经常性遭遇的冗长时延和突变时延。 这使得目前GPRS网络中使用的传统TCP或UDP传输协议均不能适应GPRS自身的链路特性，也不能实现高效的分组传输。-This paper mainly studied the W-WAN in the GPRS network for mobile video transmission problems. The article discusses the GPRS network architecture, protocol architecture, as well as link characteristics, from which we can see: GPRS network link not only has the general wireless link data transmission delay encountered in large, common problems of higher bit error rate, and with mobile communications system uses a cellular frequency reuse methods brought about by cell switching, as well as the communication link performance with the location change and change is inevitable GPRS network, the recurrent encountered lengthy delay and delay mutation. This makes the current GPRS network using the traditional TCP or UDP transport protocol are una

radius AAA 源代码,基于radius协议的鉴权，认证组件，广泛用于电信行业和银行业。-radius AAA source code, based on the radius of the agreement authentication, authentication components widely used in the telecommunications industry and banking.

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.

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

说明：网上的实现基于SIP协议通信的例子，来源于百度文库。已经经过测试，完全可以编译成功。使用平台：VS2010lib目录：库文件，eXosip2.lib, libeay32.libosip2.lib osipparser2.lib ssleay32.libinc目录：所用到库的头文件。dll 目录：动态库文件uac：用户代理客户端uas：用户代理服务器端

这是h.323交互话音应答代码-interactive voice response code

H323_CallingLineIdentificationPresentation.rar