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JS code runs on the browser and uses the browser's webrtc implementation. C code uses gstreamer's webrtc implementation, for which you need the following repositories: 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/
1.3 KiB
1.3 KiB
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
idtoo. - Run
webrtc-sendrecv --peer-id=IDwith theidfrom 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