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gstreamer/webrtc/multiparty-sendrecv/gst
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Nirbheek Chauhan 9b1a0e5389 WIP: Add a new multiparty sendrecv gstreamer demo 
You can join a room and an audio-only call will be started with all
peers in that room. Currently uses audiotestsrc only.

BUG: With >2 peers in a call, if a peer leaves, the pipeline stops
     outputting data from the remaining peers to the (audio) sink.

TODO: JS code to allow a browser to join the call
TODO: Cleanup pipeline when a peer leaves
TODO: Add ICE servers to allow calls over the Internet
TODO: Perhaps setup a TURN server as well
2017-10-30 09:14:29 +05:30
..
.gitignore WIP: Add a new multiparty sendrecv gstreamer demo 2017-10-30 09:14:29 +05:30
mp-webrtc-sendrecv.c WIP: Add a new multiparty sendrecv gstreamer demo 2017-10-30 09:14:29 +05:30