visual C#源码，摄像头捕获,视频会议，网络互通

OPENH323源代码，对于开发视频会议非常有用-OPENH323 SOURCE, useful for netmeeting

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

目前网络上面sip协议比较流行,现在这个是一个sip协议的测试软件,可以对当前的sip软件进行版本兼容性测试.-above the current network sip agreement more popular, and now this agreement is a sip of testing software that the current version sip software compatibility testing.

基于VOCAL协议栈的SIP用户代理端，可以在linux下进行SIP呼叫-invisible protocol stack based on the SIP user agent side, the under linux SIP Call

GSM 06.10 13kbit/s RPE/LTP 语音压缩算法 -GSM 06.10 13kbit/s RPE/LTP voice compression algorithm

jrtp流媒体传输的code，适合大家参考编程用-jrtp streaming media transmission of code, suitable for everyone with reference programming

IP电话会议系统，支持多人会议，分为Clien和Server两部分，并附有UML设计.-IP telephone conference systems to support people, Client and Server divided into two parts. together with UML design.

网络电话工具软件-Internet phone tool software

Python 回声取消基于 Speex lib。使用ctypes speex 与工作

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