# webrtcsink and webrtcsrc 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](signalling/src/bin/server.rs)) and a Javascript API ([gstwebrtc-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. [webrtcbin]: https://gstreamer.freedesktop.org/documentation/webrtc/index.html ## Features `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](src/signaller/mod.rs). The [default signaller](src/signaller/imp.rs) can be used as an example. An [example](examples/webrtcsink-custom-signaller/README.md) 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. [example project]: https://github.com/centricular/webrtcsink-custom-signaller ## Building > Make sure to install the development packages for some codec libraries > beforehand, such as libx264, libvpx and libopusenc, exact names depend > on your distribution. ``` shell cargo build ``` ## Usage Open three terminals. In the first one, run the signalling server: ``` shell 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): ``` shell cd gstwebrtc-api npm install npm start ``` In the third one, run a webrtcsink producer from a GStreamer pipeline: ``` shell export GST_PLUGIN_PATH=/target/debug:$GST_PLUGIN_PATH gst-launch-1.0 webrtcsink name=ws meta="meta,name=gst-stream" videotestsrc ! ws. audiotestsrc ! ws. ``` The webrtcsink produced stream will appear in the former web page (automatically opened at https://localhost:9090) under the name "gst-stream", if you click on it you should see a test video stream and hear a test tone. You can also produce WebRTC streams from the web browser and consume them with a GStreamer pipeline. Click on the "Start Capture" button and copy the "Client ID" value. Then open a new terminal and run: ``` shell export GST_PLUGIN_PATH=/target/debug:$GST_PLUGIN_PATH gst-launch-1.0 playbin uri=gstwebrtc://127.0.0.1:8443?peer-id=[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. ## Configuration 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: ``` shell gst-launch-1.0 webrtcsink signaller::uri="ws://127.0.0.1:8443" .. ``` ### 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](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: ``` shell gst-launch-1.0 wpesrc location=https://gstreamer.freedesktop.org/documentation/ ! 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 https://127.0.0.1:9090 if you followed previous steps in that readme) or with the following GSteamer pipeline as a client: ``` shell gst-launch-1.0 webrtcsrc signaller::producer-peer-id= enable-data-channel-navigation=true ! videoconvert ! autovideosink ``` [`GstNavigation`]: https://gstreamer.freedesktop.org/documentation/video/gstnavigation.html [`wpesrc`]: https://gstreamer.freedesktop.org/documentation/wpe/wpesrc.html ## 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: ``` shell $HOME/go/bin/comcast --device=$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: ``` shell $HOME/go/bin/comcast --device=$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: ``` shell gst-launch-1.0 webrtcsink congestion-control=disabled ``` [simple tool]: https://github.com/tylertreat/comcast ## Monitoring tool An example of client/server application for monitoring per-consumer stats can be found [here]. [here]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/tree/main/net/webrtc/examples ## License All the rust code in this repository is licensed under the [Mozilla Public License Version 2.0]. Code in [gstwebrtc-api](gstwebrtc-api) is also licensed under the [Mozilla Public License Version 2.0]. [Mozilla Public License Version 2.0]: http://opensource.org/licenses/MPL-2.0 ## Using the AWS KVS signaller * Setup AWS Kinesis Video Streams * Create a channel from the AWS console () * Start a producer: ``` AWS_ACCESS_KEY_ID="XXX" AWS_SECRET_ACCESS_KEY="XXX" gst-launch-1.0 videotestsrc pattern=ball ! video/x-raw, width=1280, height=720 ! videoconvert ! textoverlay text="Hello from GStreamer!" ! videoconvert ! awskvswebrtcsink name=ws signaller::channel-name="XXX" ``` * Connect a viewer @ ## Using the WHIP Signaller Testing the whip signaller can be done by setting up janus and . * Set up a [janus] instance with the videoroom plugin configured to expose a room with ID 1234 (configuration in `janus.plugin.videoroom.jcfg`) * Open the web page, click start and join the room * Set up the [simple whip server] as explained in its README * Navigate to , create an endpoint named room1234 pointing to the Janus room with ID 1234 * Finally, send a stream to the endpoint with: ``` shell gst-launch-1.0 -e uridecodebin uri=file:///home/meh/path/to/video/file ! \ videoconvert ! video/x-raw ! queue ! \ whipwebrtcsink name=ws signaller::whip-endpoint="http://127.0.0.1:7080/whip/endpoint/room1234" ``` You should see a second video displayed in the videoroomtest web page. ## Using the LiveKit Signaller Testing the LiveKit signaller can be done by setting up [LiveKit] and creating a room. You can connect either by given the API key and secret: ``` shell gst-launch-1.0 -e uridecodebin uri=file:///home/meh/path/to/video/file ! \ videoconvert ! video/x-raw ! queue ! \ livekitwebrtcsink signaller::ws-url=ws://127.0.0.1:7880 signaller::api-key=devkey signaller::secret-key=secret signaller::room-name=testroom ``` Or by using a separately created authentication token ``` shell gst-launch-1.0 -e uridecodebin uri=file:///home/meh/path/to/video/file ! \ videoconvert ! video/x-raw ! queue ! \ livekitwebrtcsink signaller::ws-url=ws://127.0.0.1:7880 signaller::auth-token=mygeneratedtoken signaller::room-name=testroom ``` You should see a second video displayed in the videoroomtest web page. [LiveKit]: https://livekit.io/ [janus]: https://github.com/meetecho/janus-gateway [simple whip server]: https://github.com/meetecho/simple-whip-server/