symbian platform simple SIP

voip电话语音信号编码，传输方式，大家顶！-voip telephone speech signal coding, transmission mode, the top U.S.!

openh323 通用库 -Universal library for openh323

Portable Unix Libary

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

基于TCP传输协议，通过wifi网络实现PC端和PPC端之间的实时视频传输，。-agreement through wifi network PC and PPC-end between the real-time video transmission.

OSIP.PDF

一个linux下的高速sip router-a sip of high-speed router

rtpbreak 将语音数据从抓取的包中提取出来.-rtpbreak voice data from the extracted package to crawl out.

第十章代码，基于ＶＩＳ　Ｈ．３２３　Ｄｌｌ　Ｌｉｂｒａｒｙ的视频会议系统-Chapter X code, based on the VIS H. 323 Dll Library of Video Conference System

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