基于 SIP 的 VOIP 电话

gsm算法最新源码，适用于语音压缩，如gsm和voip系统-gsm latest source algorithm applicable to voice compression, such as GSM and voip system

支持h323的视频电话客户端-support h323 video telephony client

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

一个基于Openh323的视频会议终端，基于VC开发环境，可以与netmeeting互联。-It is a video meeting terminal based on Openh323 protocol, which developed with VC and can communicate with netmeeting.

AppWizard has created this NetPhone application for you. This application not only demonstrates the basics of using the Microsoft Foundation classes but is also a starting point for writing your application.-AppWizard has created this NetPhone application for you. This application not only demonstrates the basics of using the Microsoft Foundation classes but is also a starting point for writing your application.

实现局域网的屏幕传输 可获取计算机的视频驱动实现高速传输 -Realize LAN can access the computer screen transmission of video drivers to achieve high-speed transmission

windows 下 一个很好用的VOIp 网守源码~~不错的-windows used in the next one very good source VOIp Gatekeeper ~ ~ good

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

Low Rate 语音压缩 -Low Rate voice compression algorithm

使用JRTPLIB库实现RTP的传输RTP是目前解决流媒体实时传输问题的最好办法，如果需要在Linux平台上进行实时流媒体编程，可以考虑使用一些开放源代码的RTP库，如LIBRTP、JRTPLIB等。JRTPLIB是一个面向对象的RTP库，它完全遵循RFC 1889设计，在很多场合下是一个非常不错的选择，下面就以JRTPLIB为例，讲述如何在Linux平台上运用RTP协议进行实时流媒体编程。-JRTPLIB use of RTP for achieving the transmission RTP is to solve real-time streaming media transmission problems the best way, if necessary on the Linux platform for real-time streaming media programming, can consider using open-source library of RTP, as LIBRTP, JRTPLIB so. JRTPLIB is an object-oriented library RTP, which fully comply with the RFC 1889 design, in many occasions is a very good choice, following on to JRTPLIB example, on how the Linux platform on the use of RTP for real-time streaming media programming.