gstreamer/webrtc

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.

http://cgit.freedesktop.org/gstreamer/gstreamer http://cgit.freedesktop.org/gstreamer/gst-plugins-base http://cgit.freedesktop.org/gstreamer/gst-plugins-good http://cgit.freedesktop.org/gstreamer/gst-plugins-bad

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