SIP Stackslibmsip: A C++ SIP stack for Linux developed for the miniSIP project....

SIP RFC协议标准,只可惜是 英文版的，可做参考-SIP RFC protocol standards, but unfortunately are in English, can refer to

asterisk 电话未来之路 一本非常有意义的数据 中文asterisk文档 对做voip 和PBX的朋友们有很大帮助-Future of asterisk phone a very meaningful data to make Chinese asterisk documents voip and PBX friends are very helpful

应用背景一个拨号应用程序，任何人（甚至老年人）可以使用。它有大的钥匙可定制任何颜色或大小是非常有用的。大量的设置让它成为你所需要的！对视力不好的人或是完美的老年人！它是很好的长号码拨号！关键技术Android手机拨号上网使用SIP的VoIP协议。它允许使用互联网的免费电话。特点：用于快速大按钮和显示，方便拨号•可选的接触按钮直接从您的联系人列表拨号•用最少的努力迅速增加新的接触号码

windows/linux平台下C语言编写的sip软电话siphon最新版本。-windows/linux platform C language prepared by the sip soft phone siphon latest version.

点对点语音通信控件V3.6，集成了G729A压缩算法，实时传输协议，话音清晰流畅！使用简单方便！适用于VC＋＋、VB、 dot NET平台、C++Builder、Delphi、易语言及网页应用程序等常见开发平台。-Point-to-point voice communication control V3.6, integrated G729A compression algorithm, Real-time Transport Protocol, voice clear and smooth! The use of simple and convenient! Apply to VC , VB, dot NET platform, C++ Builder, Delphi, easy language and procedures, and other common web application development platform.

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.

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.

2.0说明 1.0支持系统的录音多种格式,有一定的延时,没有过滤杂音功能. 2.0已取消支持系统的录音多种格式,使用PCM采集数据G711A压缩格式(8000HZ单声道16位格式录音每秒以8K完成数据,16000HZ单声道16位格式录音每秒以11K完成数据,音质相当好),延时降低到最小100-500MS以内,不会随时间增加而增加延时(如果是说话测试一直保持200MS的延时,如果是用播放歌曲来测试,有自动校正延时功能,恢复成200MS的延时,恢复过程中不会中断歌曲的播放,只是小小加快唱歌的速度,听觉不会觉察出来),加入了过滤杂音功能. 2.0安装 将ACMG711.dll复制到系统文件夹里 再安装控件 -2.0 1.0 Note recording system to support multiple formats, a certain delay, no noise filtering function. 2.0 support system has been abolished recording formats, the use of PCM data acquisition G711A compression format (8000HZ 16 mono recording format to 8K per second complete data 16000HZ mono recording format 16 to 11K per second complete data quality is very good), Delay reduced to the minimum 100-500MS within not with time delay increases (if word has been tested 200MS delay, If it is used to test play the songs, the automatic correction function of delay, 200MS resume into the delay, the recovery process will not be interrupted play songs, singing just a small speed up

语音聊天系统 采用VB 开发 包含客户端和服务器端-Voice chat system uses the VB development includes client and server

VOIP program in VC++, video, voice and other features