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(parent: 569aff4) | patch
WIP: Add a new multiparty sendrecv gstreamer demo
authorNirbheek Chauhan <nirbheek@centricular.com>
Mon, 30 Oct 2017 03:39:36 +0000 (09:09 +0530)
committerNirbheek Chauhan <nirbheek@centricular.com>
Mon, 30 Oct 2017 03:44:29 +0000 (09:14 +0530)
commit9b1a0e538992444b292f8bbca38bbe4a7da4c7ab
tree0786d65829f465c1157c67c56fb4d05549292902tree | snapshot
parent569aff43f9e80f95eb6ce6e7de190b9664775b00commit | diff
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
webrtc/multiparty-sendrecv/gst/.gitignore [new file with mode: 0644] blob
webrtc/multiparty-sendrecv/gst/mp-webrtc-sendrecv.c [new file with mode: 0644] blob
Domain: Multimedia / Media Common;