6.3 KiB
gstwebrtc-api
Javascript API used to integrate GStreamer WebRTC streams produced and consumed by webrtcsink and webrtcsrc elements into a web browser or a mobile WebView.
This API allows a complete 360º interconnection between GStreamer and web interfaces for realtime streaming using the WebRTC protocol.
This API is released under the Mozilla Public License Version 2.0 (MPL-2.0) that can be found in the LICENSE-MPL-2.0 file or at https://opensource.org/licenses/MPL-2.0
Copyright (C) 2022 Igalia S.L. <info@igalia.com>
Author: Loïc Le Page <llepage@igalia.com>
It includes external source code from webrtc-adapter that is embedded with the API. The webrtc-adapter BSD 3-Clause license is available at https://github.com/webrtcHacks/adapter/blob/master/LICENSE.md
Webrtc-adapter is Copyright (c) 2014, The WebRTC project authors, All rights reserved.
Copyright (c) 2018, The adapter.js project authors, All rights reserved.
It also includes Keyboard.js source code from the Apache guacamole-client that is embedded with the API to enable remote control of the webrtcsink producer.
Keyboard.js is released under the Apache 2.0 license available at: https://github.com/apache/guacamole-client/blob/master/LICENSE
Building the API
The GstWebRTC API uses Webpack to bundle all source files and dependencies together.
You only need to install Node.js to run all commands.
On first time, install the dependencies by calling:
$ npm install
Then build the bundle by calling:
$ npm run make
It will build and compress the code into the dist/ folder, there you will find 2 files:
- gstwebrtc-api-[version].min.js which is the only file you need to include into your web application to use the API. It already embeds all dependencies.
- gstwebrtc-api-[version].min.js.map which is useful for debugging the API code, you need to put it in the same folder as the API script on your web server if you want to allow debugging, else you can just ignore it.
The API documentation is created into the docs/ folder. It is automatically created when building the whole API.
If you want to build the documentation only, you can call:
$ npm run docs
If you only want to build the API without the documentation, you can call:
$ npm run build
Packaging the API
You can create a portable package of the API by calling:
$ npm pack
It will create a gstwebrtc-api-[version].tgz file that contains all source code, documentation and built API. This portable package can be installed as a dependency in any Node.js project by calling:
$ npm install gstwebrtc-api-[version].tgz
Testing and debugging the API
To easily test and debug the GstWebRTC API, you just need to:
- launch the webrtc signalling server by calling (from the repository gst-plugins-rs root folder):
$ cargo run --bin gst-webrtc-signalling-server - launch the GstWebRTC API server by calling (from the net/webrtc/gstwebrtc-api sub-folder):
$ npm start
It will launch a local HTTPS server listening on port 9090 and using an automatically generated self-signed certificate.
With this server you can test the reference example shipped in index.html from a web browser on your local computer or a mobile device.
Interconnect with GStreamer pipelines
Once the signalling and gstwebrtc-api servers launched, you can interconnect the streams produced and consumed from the web browser with GStreamer pipelines using the webrtcsink and webrtcsrc elements.
Consume a WebRTC stream produced by the gstwebrtc-api
On the web browser side, click on the Start Capture button and give access to the webcam. The gstwebrtc-api will start producing a video stream.
The signalling server logs will show the registration of a new producer with the same peer_id as the Client ID that appears on the webpage.
Then launch the following GStreamer pipeline:
$ gst-launch-1.0 playbin uri=gstwebrtc://[signalling server]?peer-id=[client ID of the producer]
Using the local signalling server, it will look like this:
$ gst-launch-1.0 playbin uri=gstwebrtc://127.0.0.1:8443?peer-id=e54e5d6b-f597-4e8f-bc96-2cc3765b6567
The underlying uridecodebin element recognizes the gstwebrtc:// scheme as a WebRTC stream compatible with the gstwebrtc-api and will correctly use a webrtcsrc element to manage this stream.
The gstwebrtc:// scheme is used for normal WebSocket connections to the signalling server, and the gstwebrtcs:// scheme for secured connections over SSL or TLS.
Produce a GStreamer WebRTC stream consumed by the gstwebrtc-api
Launch the following GStreamer pipeline:
$ gst-launch-1.0 videotestsrc ! agingtv ! webrtcsink meta="meta,name=native-stream"
By default webrtcsink element uses ws://127.0.0.1:8443 for the signalling server address, so there is no need
for more arguments. If you're hosting the signalling server elsewhere, you can specify its address by adding
signaller::uri="ws[s]://[signalling server]" to the list of webrtcsink properties.
Once the GStreamer pipeline launched, you will see the registration of a new producer in the logs of the signalling server and a new remote stream, with the name native-stream, will appear on the webpage.
You just need to click on the corresponding entry to connect as a consumer to the remote native stream.
Produce a GStreamer interactive WebRTC stream with remote control
Launch the following GStreamer pipeline:
$ gst-launch-1.0 wpesrc location=https://gstreamer.freedesktop.org/documentation ! queue ! webrtcsink enable-data-channel-navigation=true meta="meta,name=web-stream"
Once the GStreamer pipeline launched, you will see a new producer with the name web-stream. When connecting to this producer you will see the remote rendering of the web page. You can interact remotely with this web page, controls are sent through a special WebRTC data channel while the rendering is done remotely by the wpesrc element.