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Error to trace to log to deploy. One click. No SSH.
Catch the cause before the pager goes off.
AppSignal links every error to the trace, the trace to the log, the log to the deploy that shipped it.
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