{"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":743,"stargazers":743,"forks":346,"fork":false,"size":8138,"owner":"muaz-khan","private":false,"open_issues":74,"has_issues":true,"has_downloads":true,"has_wiki":true,"language":"JavaScript","created_at":"2012-11-13T21:53:13-08:00","pushed_at":"2013-09-11T09:40:05-07:00","master_branch":"master"},"actor_attributes":{"type":"User","name":"amoa400","email":"030ecd4983cbf16841ebd678583e55371be2966a@gmail.com"},"url":"https://github.com/amoa400/WebRTC-Experiment"}