GStreamer WebRTC demos

All demos use the same signalling server in the signalling/ directory

Downloading GStreamer

The GStreamer WebRTC implementation has now been merged upstream, and is in the GStreamer 1.14 release. Binaries can be found here:

https://gstreamer.freedesktop.org/download/

Building GStreamer from source

If you don't want to use the binaries provided by GStreamer or on your Linux distro, you can build GStreamer from source.

The easiest way to build the webrtc plugin and all the plugins it needs, is to use Cerbero. These instructions should work out of the box for all platforms, including cross-compiling for iOS and Android.

One thing to note is that it's written in Python 2, so you may need to replace all instances of ./cerbero-uninstalled (or cerbero) with python2 cerbero-uninstalled or whatever Python 2 is called on your platform.

Building GStreamer manually from source

For hacking on the webrtc plugin, you may want to build manually using the git repositories:

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/

You may need to install the following packages using your package manager:

json-glib, libsoup, libnice, libnice-gstreamer1 (the gstreamer plugin for libnice)

Filing bugs

Please only file bugs about the demos here. Bugs about GStreamer's WebRTC implementation should be filed on the GStreamer bugzilla.

You can also find us on IRC by joining #gstreamer @ FreeNode.

Documentation

Currently, the best way to understand the API is to read the examples. This post breaking down the API should help with that:

http://blog.nirbheek.in/2018/02/gstreamer-webrtc.html

Examples

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 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

TODO: Port to Python and Rust.

multiparty-sendrecv: Multiparty audio conference with N peers

Build the sources in the gst/ directory on your machine

$ 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

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.
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.
- To change what a peer sends, find the audiotestsrc element in the source and change the wave property.
- You can, of course, also replace audiotestsrc itself with autoaudiosrc (any platform) or pulsesink (on linux).
TODO: implement JS to do the same, derived from the JS for the sendrecv example.

TODO: Selective Forwarding Unit (SFU) example

Server routes media between peers
Participant sends 1 stream, receives n-1 streams

TODO: Multipoint Control Unit (MCU) example

Server mixes media from all participants
Participant sends 1 stream, receives 1 stream

shadow1runner/gstwebrtc-demos