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gstreamer/webrtc
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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
..
multiparty-sendrecv/gst WIP: Add a new multiparty sendrecv gstreamer demo 2017-10-30 09:14:29 +05:30
sendrecv sendrecv: Rename function for greater clarity 2017-10-30 09:14:29 +05:30
signalling Update Protocol.md 2017-10-29 04:08:45 +05:30
.gitignore Add sendrecv implementation in js and gst webrtc 2017-10-21 20:02:19 +05:30
LICENSE Initial commit 2017-10-21 19:43:01 +05:30
README.md Add sendrecv implementation in js and gst webrtc 2017-10-21 20:02:19 +05:30

README.md

GStreamer WebRTC demos

All demos use the same signalling server in the signalling/ directory

You will need the following repositories till the GStreamer WebRTC implementation is merged upstream:

https://github.com/ystreet/gstreamer/tree/promise

https://github.com/ystreet/gst-plugins-bad/tree/webrtc

You can build these with either Autotools gst-uninstalled:

https://arunraghavan.net/2014/07/quick-start-guide-to-gst-uninstalled-1-x/

Or with Meson gst-build:

https://cgit.freedesktop.org/gstreamer/gst-build/

sendrecv: Send and receive audio and video

  • Serve the js/ directory on the root of your website, or open https://webrtc.nirbheek.in
    • The JS code assumes the signalling server is on port 8443 of the same server serving the HTML
  • Build and run the sources in the gst/ directory on your machine
$ gcc webrtc-sendrecv.c $(pkg-config --cflags --libs gstreamer-webrtc-1.0 gstreamer-sdp-1.0 libsoup-2.4 json-glib-1.0) -o webrtc-sendrecv
  • Open the website in a browser and ensure that the status is "Registered with server, waiting for call", and note the id too.
  • Run webrtc-sendrecv --peer-id=ID with the id from the browser. You will see state changes and an SDP exchange.
  • You will see a bouncing ball + hear red noise in the browser, and your browser's webcam + mic in the gst app