procedure is a common source! !

VOIP(H323)源码，实现了G723.1 G729 G728 G711等音频算法，可以和NetMeeting互通-VoIP (H323) source, Implementation of G723.1 G729 G728 G711 audio algorithms, etc., NetMeeting and can exchange

openh323 source code

vc编写的sip代理服务器,无需其他的库，直接编译-vc prepared sip proxy server, no additional library is directly compiled

CS多客户端聊天工具 V1.0 可以多人在线聊天。经测试可以三十人以上，更多未测。

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

STUN，VOIP中用来穿透NAT的一种方式，可以穿过基本的三种NAT-STUN, VoIP NAT used to penetrate a way, through the three basic NAT

视频会议web系统-web video conference system

Portable Windows Libary

RFC文档中有关SIP协议的，3261，3264，3428，3265，2327，2543，大部分都是中文的，还有中华人民共和国通信行业标准中有关SIP的标准。-RFC documentation of the SIP, 3261,3264,3428,3265,2327. 2,543, most of whom are Chinese, there is the People"s Republic of China telecommunications industry standards on the SIP standard.

电子白板实现，java实现，欢迎指正，-Whiteboard realize, java realize, please correct me,