SKYPE SDK 简单演示说明例程，起到抛转引玉作用

我自己开发的一个点对点的视频传输会议程序，希望大家指导呀！-I developed a point-to-point video transmission of the proceedings, we hope to guide!

JRTP最新源码 不错的RTP实现 IETF RFC 3550.-The latest source JRTP good RTP realize IETF RFC 3550.

最新的jrtplib版本,多媒体实时流传输协议（RTP/RTCP）C++ 了lib包平台要求(linux,unix,windows)用于网络音频视频传输 -Jrtplib the latest version of real-time streaming multimedia transport protocol (RTP/RTCP) C++ The lib package require platform (linux, unix, windows) for network audio video transmission

自己团队开发基于MYSQL数据库的连H323的GK的实时扣除通话用户费用的工程-their team-based MYSQL database links H323 of the GK calls deducting the cost of the project users

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

网络电话。采用GSM语音格式，2.2KB/s的数据流量，支持HTTP, Socks4/5单一代理或二者同时代理。本版本修复了以前版本的一些小问题，增加了动态调整延时，打开、关闭话筒和声音，保存和自动加载通话地址列表，代理服务器列表，使用全部或部分本机地址开机-telephone network. GSM voice format, 2.2KB/s data traffic, supports HTTP, Socks4/5 of a single agent or combination of both agents. This version fixes previous versions of some small problem, the dynamic adjustment of delay, opening, closing and microphone voice, preservation and automatically load calls address list, the proxy list, the use of all or part of the machine addresses boot

rtp relay,SIP中穿越NAT的一种方式，这是源代码，已经在red hat linux 9中调通过-rtp relay, cross SIP NAT a way, this is the source code, in red hat linux 9 emphasized through

这是voip关守的程序源代码-This is voip gatekeeper of source code

SIP Stacks libmsip: A C++ SIP stack for Linux developed for the miniSIP project. Homepage: http://www.minisip.org/libmsip/. -SIP Stackslibmsip: A C++ SIP stack for Linux developed for the miniSIP project. Homepage: http://www.minisip.org/libmsip/.

ITU-T编写的h.263协议文档，相当权威！对做视频编码解码有帮助！-ITU-T H.263 agreement prepared by the document, rather authoritative! Done right video codec help!