gstreamer

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History

Nirbheek Chauhan e9b0656bad Add sendrecv implementation in js and gst webrtc 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/		2017-10-21 20:02:19 +05:30
..
sendrecv	Add sendrecv implementation in js and gst webrtc	2017-10-21 20:02:19 +05:30
signalling	Add a simple python3 webrtc signalling server	2017-10-21 19:56:52 +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

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:

You can build these with either Autotools gst-uninstalled:

Or with Meson gst-build:

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