资源列表
conferencexp-65635
- 多媒体会议,RTP,C#,点对点,广播,-Multimedia conferencing, RTP, C#, point to point, radio,
lib
- 开发视频会话必备的开源库,已经编译好,可直接使用,有resiprocate,ortp和ffmpeg -Development of open source library video session must have been compiled, and can be used directly, there resiprocate, ortp and ffmpeg
opal-3.6.8
- opal 很多视频通话软件都会基于这种协议-opal many video call software will be based on this protocol
NetCall
- 用VC++6.0写的网络语音电话,可一对一进行语言聊天,亮点是实现了异形的图形窗口,并提供了一套该方法的模版。-Written in VC++6.0 using VoIP, you can chat one on one for language, the highlight is to achieve a shaped graphics window, and provides a template of the method.
jrtp_test
- rtp的发送端和接收端实现。需要的可以参考一下。-rtp the sender and receiver side implementation. Needed for reference.
rtspRequest
- rtsp初学者可以参考。 但是是基于ipv6的。-rtsp beginners can refer to it. But it is based on ipv6.
DemoServerSIP
- FreeSwitch server SIP demo, Visual Studio
libeXosip2-3.6.0.tar
- 开发SIP软电话的开源扩展协议库3.60版本。这是2011年10月最新版本!-Development of open source SIP soft phone extension protocol library. This is the latest version in October 2011!
pjsua
- PJSIP中的PJSUA和Simple_pjsua例程,PJSUA使用PJSIP协议栈实现了软电话的几乎所有功能,包括注册、拨打、回应等等。simple_pjsua在不到200行的代码中实现了最简单的拨打和接听电话功能-PJSIP in PJSUA and Simple_pjsua routines, PJSUA use PJSIP protocol stack to achieve a soft phone almost all functions, including registratio
gsm-1.0pl2.tar.gz
- GSM 06.10 13 kbit/s RPE/LTP语音压缩,GSM 06.1013 kbit/S RPE/LTP voice compression
IPVC.rar
- IP电视电话会议+VC源码,非常好的IP电话会议的学习代码,IP video and telephone conference on+ VC source, very good IP telephone conference learning code
videomeeting.rar
- 视频会议系统,DIRECTSHOW和VC实现,Video Conference System, DIRECTSHOW and VC to achieve