All-batteries included GStreamer WebRTC producer and consumer, that try their best to do The Right Thing™.

It also provides a flexible and all-purposes WebRTC signalling server (gst-webrtc-signalling-server) and a Javascript API (gstwebrtc-api) to produce and consume compatible WebRTC streams from a web browser.

Use case

The webrtcbin element in GStreamer is extremely flexible and powerful, but using it can be a difficult exercise. When all you want to do is serve a fixed set of streams to any number of consumers, webrtcsink (which wraps webrtcbin internally) can be a useful alternative.


webrtcsink implements the following features:

  • Built-in signaller: when using the default signalling server, this element will perform signalling without requiring application interaction. This makes it usable directly from gst-launch.

  • Application-provided signalling: webrtcsink can be instantiated by an application with a custom signaller. That signaller must be a GObject, and must implement the Signallable interface as defined here. The default signaller can be used as an example.

    An example is also available to use as a boilerplate for implementing and using a custom signaller.

  • Sandboxed consumers: when a consumer is added, its encoder / payloader / webrtcbin elements run in a separately managed pipeline. This provides a certain level of sandboxing, as opposed to having those elements running inside the element itself.

    It is important to note that at this moment, encoding is not shared between consumers. While this is not on the roadmap at the moment, nothing in the design prevents implementing this optimization.

  • Congestion control: the element leverages transport-wide congestion control feedback messages in order to adapt the bitrate of individual consumers' video encoders to the available bandwidth.

  • Configuration: the level of user control over the element is slowly expanding, consult gst-inspect-1.0 for more information on the available properties and signals.

  • Packet loss mitigation: webrtcsink now supports sending protection packets for Forward Error Correction, modulating the amount as a function of the available bandwidth, and can honor retransmission requests. Both features can be disabled via properties.

It is important to note that full control over the individual elements used by webrtcsink is not on the roadmap, as it will act as a black box in that respect, for example webrtcsink wants to reserve control over the bitrate for congestion control.

A signal is now available however for the application to provide the initial configuration for the encoders webrtcsink instantiates.

If more granular control is required, applications should use webrtcbin directly, webrtcsink will focus on trying to just do the right thing, although it might expose more interfaces to guide and tune the heuristics it employs.


Make sure to install the development packages for some codec libraries beforehand, such as libx264, libvpx and libopusenc, exact names depend on your distribution.

cargo build


Open three terminals. In the first one, run the signalling server:

cd signalling
WEBRTCSINK_SIGNALLING_SERVER_LOG=debug cargo run --bin gst-webrtc-signalling-server

In the second one, run a web browser client (can produce and consume streams):

cd gstwebrtc-api
npm install
npm start

In the third one, run a webrtcsink producer from a GStreamer pipeline:

export GST_PLUGIN_PATH=<path-to-gst-plugins-rs>/target/debug:$GST_PLUGIN_PATH
gst-launch-1.0 webrtcsink name=ws meta="meta,name=gst-stream" videotestsrc ! ws. audiotestsrc ! ws.

The webrtcsink produced stream will appear in the former web page (automatically opened at https://localhost:9090) under the name "gst-stream", if you click on it you should see a test video stream and hear a test tone.

You can also produce WebRTC streams from the web browser and consume them with a GStreamer pipeline. Click on the "Start Capture" button and copy the "Client ID" value.

Then open a new terminal and run:

export GST_PLUGIN_PATH=<path-to-gst-plugins-rs>/target/debug:$GST_PLUGIN_PATH
gst-launch-1.0 playbin uri=gstwebrtc://[Client ID]

Replacing the "peer-id" value with the previously copied "Client ID" value. You should see the playbin element opening a window and showing you the content produced by the web page.


The webrtcsink element itself can be configured through its properties, see gst-inspect-1.0 webrtcsink for more information about that, in addition the default signaller also exposes properties for configuring it, in particular setting the signalling server address, those properties can be accessed through the gst::ChildProxy interface, for example with gst-launch:

gst-launch-1.0 webrtcsink signaller::uri="ws://" ..

Enable 'navigation' a.k.a user interactivity with the content

webrtcsink implements the GstNavigation interface which allows interacting with the content, for example move with your mouse, entering keys with the keyboard, etc... On top of that a WebRTCDataChannel based protocol has been implemented and can be activated with the enable-data-channel-navigation=true property allowing a client to send GstNavigation events using the WebRTC data channel.

The gstwebrtc-api and webrtcsrc implement the protocol as well and they can be used as a client to control a remote sever.

You can easily test this feature using the wpesrc element with the following pipeline that will start a server that allows you to navigate the GStreamer documentation:

gst-launch-1.0 wpesrc location= ! queue ! webrtcsink enable-data-channel-navigation=true meta="meta,name=web-stream"

You can control it inside the video running within your web browser (at if you followed previous steps in that readme) or with the following GSteamer pipeline as a client:

gst-launch-1.0 webrtcsrc signaller::producer-peer-id=<webrtcsink-peer-id> enable-data-channel-navigation=true ! videoconvert ! autovideosink

Sending HTTP headers

During the initial signalling server handshake, you have the option to transmit HTTP headers, which can be utilized, for instance, for authentication purposes or sticky sessions:

gst-launch-1.0 webrtcsink signaller::uri="ws://" signaller::headers="headers,foo=bar,cookie=\"session=1234567890; foo=bar\""

Testing congestion control

For the purpose of testing congestion in a reproducible manner, a simple tool has been used, it has been used on Linux exclusively but it is also documented as usable on MacOS too. Client web browser has to be launched on a separate machine on the LAN to test for congestion, although specific configurations may allow to run it on the same machine.

Testing procedure was:

  • identify the server machine network interface (e.g. with ifconfig on Linux)

  • identify the client machine IP address (e.g. with ifconfig on Linux)

  • start the various services as explained in the Usage section (use GST_DEBUG=webrtcsink:7 to get detailed logs about congestion control)

  • start playback in the client browser

  • Run a comcast command on the server machine, for instance:

    $HOME/go/bin/comcast --device=$SERVER_INTERFACE --target-bw 3000 --target-addr=$CLIENT_IP --target-port=1:65535 --target-proto=udp
  • Observe the bitrate sharply decreasing, playback should slow down briefly then catch back up

  • Remove the bandwidth limitation, and observe the bitrate eventually increasing back to a maximum:

    $HOME/go/bin/comcast --device=$SERVER_INTERFACE --stop

For comparison, the congestion control property can be set to "disabled" on webrtcsink, then the above procedure applied again, the expected result is for playback to simply crawl down to a halt until the bandwidth limitation is lifted:

gst-launch-1.0 webrtcsink congestion-control=disabled

Monitoring tool

An example of client/server application for monitoring per-consumer stats can be found here.


All the rust code in this repository is licensed under the [Mozilla Public License Version 2.0].

(from gst-plugin-webrtc)
Name Classification Description
awskvswebrtcsink Sink/Network/WebRTC WebRTC sink with kinesis video streams signaller
janusvrwebrtcsink Sink/Network/WebRTC WebRTC sink with Janus Video Room signaller
livekitwebrtcsink Sink/Network/WebRTC WebRTC sink with LiveKit signaller
livekitwebrtcsrc Source/Network/WebRTC WebRTC source with LiveKit signaller
webrtcsink Sink/Network/WebRTC WebRTC sink with custom protocol signaller
webrtcsrc Source/Network/WebRTC WebRTC src
whipclientsink Sink/Network/WebRTC WebRTC sink with WHIP client signaller
whipserversrc Source/Network/WebRTC WebRTC source element using WHIP Server as the signaller


awskvswebrtcsink – WebRTC sink with kinesis video streams signaller



janusvrwebrtcsink – WebRTC sink with Janus Video Room signaller

livekitwebrtcsink – WebRTC sink with LiveKit signaller

livekitwebrtcsrc – WebRTC source with LiveKit signaller

webrtcsink – WebRTC sink with custom protocol signaller

webrtcsrc – WebRTC src

whipclientsink – WebRTC sink with WHIP client signaller

whipserversrc – WebRTC source element using WHIP Server as the signaller

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