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Tim-Philipp Müller 2e5204ae3b README: fix formatting
2018-02-02 08:41:21 +00:00
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multiparty-sendrecv/gst Update to new promise API 2017-11-22 22:28:55 +10:00
sendrecv webrtc-sendrecv: define GST_USE_UNSTABLE_API to avoid compiler warnings 2018-02-02 08:39:04 +00:00
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 README: fix formatting 2018-02-02 08:41:21 +00:00

README.md

GStreamer WebRTC demos

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

The GStreamer WebRTC implementation has now been merged upstream, so all you need is the latest GStreamer git master, as of 2 February 2018 or later:

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