{"repository":{"description":"WebRTC audio/video conferencing, audio/video recording, screen-sharing, file-sharing, WebRTC data-channels, pre-recorded media streaming, part-of-screen sharing, broadcasting, hangout; WebRTC signaling: SIP, WebSockets, socket.io; peer-to-peer and peer-to-server based WebRTC Experiments. Group text-chat and file-transferring; audio-only calls; and much more! WebRTC libraries like RecordRTC, RTCPeerConnection, RTCMultiConnection, RTCall, and DataChannel.js ","homepage":"https://www.webrtc-experiment.com/","watchers":960,"stargazers":960,"forks":437,"fork":false,"size":8455,"owner":"muaz-khan","private":false,"open_issues":88,"has_issues":true,"has_downloads":true,"has_wiki":true,"language":"JavaScript","created_at":"2012-11-13T21:53:13-08:00","pushed_at":"2013-12-19T23:02:00-08:00","master_branch":"master"},"actor_attributes":{"type":"User","email":"da39a3ee5e6b4b0d3255bfef95601890afd80709"},"url":"https://github.com/muaz-khan/WebRTC-Experiment"}