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32 lines
1.3 KiB
Markdown
32 lines
1.3 KiB
Markdown
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## GStreamer WebRTC demos
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All demos use the same signalling server in the `signalling/` directory
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You will need the following repositories till the GStreamer WebRTC implementation is merged upstream:
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https://github.com/ystreet/gstreamer/tree/promise
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https://github.com/ystreet/gst-plugins-bad/tree/webrtc
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You can build these with either Autotools gst-uninstalled:
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https://arunraghavan.net/2014/07/quick-start-guide-to-gst-uninstalled-1-x/
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Or with Meson gst-build:
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https://cgit.freedesktop.org/gstreamer/gst-build/
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### sendrecv: Send and receive audio and video
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* Serve the `js/` directory on the root of your website, or open https://webrtc.nirbheek.in
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- The JS code assumes the signalling server is on port 8443 of the same server serving the HTML
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* Build and run the sources in the `gst/` directory on your machine
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```console
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$ 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
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```
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* Open the website in a browser and ensure that the status is "Registered with server, waiting for call", and note the `id` too.
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* Run `webrtc-sendrecv --peer-id=ID` with the `id` from the browser. You will see state changes and an SDP exchange.
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* You will see a bouncing ball + hear red noise in the browser, and your browser's webcam + mic in the gst app
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