webrtcsink

All-batteries included GStreamer WebRTC producer, that tries its best to do The Right Thing™.

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.

Features

webrtcsink implements the following features:

• Built-in signaller: when using the default signalling server (provided as a python script here), 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.

• 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.

• Configuration: the level of user control over the element is at the moment quite narrow, as the only interface exposed is control over proposed codecs, as well as their order of priority. Consult gst-inspect=1.0 for more information.

More features are on the roadmap, focusing on mechanisms for mitigating packet loss and congestion.

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.

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 interfaces to guide and tune the heuristics it employs.

Building

Prerequisites

The element has only been tested for now against GStreamer master, with an extra Merge Request pending review:

https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1233

The MR should hopefully make it in in time for GStreamer's 1.20 release, in the meantime the patches must be applied locally.

For testing, it is recommended to simply build GStreamer locally and run in the uninstalled devenv.

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

git clone https://gitlab.freedesktop.org/meh/gstreamer/-/tree/webrtcsink
meson build
ninja -C build
ninja -C build devenv

Compiling

cargo build

Usage

Open three terminals. In the first, run:

cd signalling
python3 simple-server.py --addr=127.0.0.1 --disable-ssl

In the second, run:

cd www
python3 -m http.server

In the third, run:

export GST_PLUGIN_PATH=$PWD/target/debug:$GST_PLUGIN_PATH
gst-launch-1.0 webrtcsink name=ws videotestsrc ! ws. audiotestsrc ! ws.

When the pipeline above is running succesfully, open a browser and point it to the python server:

xdg-open http://127.0.0.1:8000

You should see an identifier listed in the left-hand panel, click on it. You should see a test video stream, and hear a test tone.

License

All code in this repository is licensed under the MIT license.