WebRTC audio/video call and conferencing server.
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Joachim Bauch 44e666f2c8
Handle gUM failures on FF when no audio or video device is available.
9 years ago
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api.js Implement TURN service client 10 years ago
connector.js Bumped copyright to 2015. 11 years ago
dummystream.js Improved is DummyStream check. 11 years ago
peercall.js Explicitly pass "outgoing" flag when creating PeerCall object. 9 years ago
peerconference.js Try to recover from (some) lost p2p connection state. 10 years ago
peerconnection.js Major refactoring of call / conference handling. 10 years ago
peerscreenshare.js Moved from offerConstraints to offerOptions which follows latest spec. This fixes offer creation in Firefox >=43. 10 years ago
peerxfer.js Moved from offerConstraints to offerOptions which follows latest spec. This fixes offer creation in Firefox >=43. 10 years ago
tokens.js Fixed jshint latedef and strict errors. 10 years ago
usermedia.js Handle gUM failures on FF when no audio or video device is available. 9 years ago
utils.js Filter rtx support from remote SDP for Chrone <= 38 10 years ago
webrtc.js Explicitly pass "outgoing" flag when creating PeerCall object. 9 years ago