gstreamer/webrtc/multiparty-sendrecv/gst/.gitignore at 180e1ce24c7742b983f0f662ab2d0421988272e3 - mirrors/gstreamer - Code: Hack the planet!

mirrors/gstreamer

mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-13 12:51:16 +00:00

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

1 line

19 B

Text

Raw Blame History

mp-webrtc-sendrecv