WebRTC audio/video call and conferencing server.
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Simon Eisenmann 2d99116aa1 Make sure to stop stream after testing user media access. 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 Fixed FireFox issue which showed multiple videos for the same peer. 11 years ago
peerconference.js Changed name of project to Spreed WebRTC. 12 years ago
peerconnection.js Fixed typo. 11 years ago
peerscreenshare.js Added settings to control audio and video constraints. Some settings were added as experiments which are disabled per default). 11 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 Make sure to stop stream after testing user media access. 11 years ago
utils.js Changed name of project to Spreed WebRTC. 12 years ago
webrtc.js Added settings to control audio and video constraints. Some settings were added as experiments which are disabled per default). 11 years ago