2018-02-17 02:40:59 +00:00
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# GStreamer WebRTC demos
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2017-10-21 14:27:29 +00:00
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All demos use the same signalling server in the `signalling/` directory
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2018-02-17 02:40:59 +00:00
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## Build
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2018-02-02 08:23:30 +00:00
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The GStreamer WebRTC implementation has now been merged upstream, so all
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2018-02-02 08:41:21 +00:00
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you need is the latest GStreamer git master, as of 2 February 2018 or later:
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2017-10-21 14:27:29 +00:00
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2018-02-02 08:41:21 +00:00
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- http://cgit.freedesktop.org/gstreamer/gstreamer
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- http://cgit.freedesktop.org/gstreamer/gst-plugins-base
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- http://cgit.freedesktop.org/gstreamer/gst-plugins-good
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- http://cgit.freedesktop.org/gstreamer/gst-plugins-bad
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2017-10-21 14:27:29 +00:00
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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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2018-02-17 02:40:59 +00:00
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You may need to install the following packages using your package manager:
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json-glib, libsoup, libnice, libnice-gstreamer1 (the gstreamer plugin for libnice)
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## Documentation
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Currently, the best way to understand the API is to read the examples. This post breaking down the API should help with that:
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http://blog.nirbheek.in/2018/02/gstreamer-webrtc.html
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## Examples
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2017-10-21 14:27:29 +00:00
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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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2018-02-17 02:40:59 +00:00
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* Build the sources in the `gst/` directory on your machine
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2017-10-21 14:27:29 +00:00
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```console
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$ gcc webrtc-sendrecv.c $(pkg-config --cflags --libs gstreamer-webrtc-1.0 gstreamer-sdp-1.0 libsoup-2.4 json-glib-1.0) -o webrtc-sendrecv
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```
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* Open the website in a browser and ensure that the status is "Registered with server, waiting for call", and note the `id` too.
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* Run `webrtc-sendrecv --peer-id=ID` with the `id` from the browser. You will see state changes and an SDP exchange.
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* You will see a bouncing ball + hear red noise in the browser, and your browser's webcam + mic in the gst app
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2018-02-17 02:40:59 +00:00
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TODO: Port to Python and Rust.
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### multiparty-sendrecv: Multiparty audio conference with N peers
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* Build the sources in the `gst/` directory on your machine
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```console
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$ gcc mp-webrtc-sendrecv.c $(pkg-config --cflags --libs gstreamer-webrtc-1.0 gstreamer-sdp-1.0 libsoup-2.4 json-glib-1.0) -o mp-webrtc-sendrecv
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```
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* Run `mp-webrtc-sendrecv --room-id=ID` with `ID` as a room name. The peer will connect to the signalling server and setup a conference room.
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* Run this as many times as you like, each will spawn a peer that sends red noise and outputs the red noise it receives from other peers.
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- To change what a peer sends, find the `audiotestsrc` element in the source and change the `wave` property.
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- You can, of course, also replace `audiotestsrc` itself with `autoaudiosrc` (any platform) or `pulsesink` (on linux).
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* TODO: implement JS to do the same, derived from the JS for the `sendrecv` example.
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### TODO: Selective Forwarding Unit (SFU) example
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* Server routes media between peers
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* Participant sends 1 stream, receives n-1 streams
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### TODO: Multipoint Control Unit (MCU) example
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* Server mixes media from all participants
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* Participant sends 1 stream, receives 1 stream
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