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SIP Video Multiconference Media Server with WebRTC support.
REPOSITORY MOVED TO GITHUB!!
https://github.com/medooze/media-server
Video Multiconference Media Server with WebRTC support.
Provide Multiconference and video broadcasting services to any SIP service.
Supports VP8, H264, MP4V-ES, H263 and H263P, continuous presence, RTMP flash broadcasting, adhoc conferences, load balancing and administrative WEB interface.
JSR309 driver implementation under development.
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A suite of open-source tools and frameworks for creating SIP apps
Session Initialtion Protocol (SIP) is widely used for telephone services over the Internet. CafeSip provides a suite of open-source tools and applications for creating customized SIP services and applications using the Java.
The G.O.N.E. is a softphone (or soft phone) running over the web, fully multi-plataform, it implements the SIP protocol, and is built to work on any SIP server, like Asterisk, and others. GONE will work on a complex sistem, but this will be showed a bit