WebRTC audio/video call and conferencing server.
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Simon Eisenmann d008ea48eb Added missing gear to remove streams. 11 years ago
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api.js Fixed call state resurrect on heartbeat timeout reconnects. 12 years ago
connector.js Fixed call state resurrect on heartbeat timeout reconnects. 12 years ago
peercall.js Enabled logging for negotiationNeeded events. 12 years ago
peerconference.js Changed name of project to Spreed WebRTC. 12 years ago
peerconnection.js Added missing gear to remove streams. 11 years ago
peerscreenshare.js Fixed typo. 12 years ago
peerxfer.js Changed name of project to Spreed WebRTC. 12 years ago
tokens.js Changed name of project to Spreed WebRTC. 12 years ago
usermedia.js Added missing gear to remove streams. 11 years ago
utils.js Changed name of project to Spreed WebRTC. 12 years ago
webrtc.js Implemented timeout for getUserMedia check. 12 years ago