# 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://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/tree/main/net/webrtc/examples/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 npm --prefix gstwebrtc-api/ run build ``` ## Usage (embedded services) `webrtcsink` can optionally instantiate a signalling server and a web server. This is the simplest set up for testing, but may not always be desirable. For instance one may prefer hosting the services on different machines, or would prefer that a crash from the host webrtcsink doesn't take down signalling / websites. Head over to the following section if you want to learn how to run services individually. In the terminal, from the root of the `net/webrtc` crate: ``` gst-launch-1.0 videotestsrc ! webrtcsink run-signalling-server=true run-web-server=true ``` In your browser of choice, navigate to , and click on the stream identifier under "Remote streams". You should see a test video stream and hear a test tone. ## Usage (standalone services) 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-control-data-channel=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-control-data-channel=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-control-data-channel=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: ``` shell gst-launch-1.0 webrtcsink signaller::uri="ws://127.0.0.1:8443" signaller::headers="headers,foo=bar,cookie=\"session=1234567890; foo=bar\"" ``` [`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 ## Control data channel When the control data channel is enabled using the `enable-control-data-channel` property, `webrtcsink` can optionally forward serialized metas and events to the consumer. Currently only timecode metas are supported, but other metas and events can easily be added in a backward-compatible manner. Here is an example for forwarding timecodes: ``` gst-launch-1.0 videotestsrc ! timecodestamper ! webrtcsink forward-metas="timecode" name=ws enable-control-data-channel=true run-signalling-server=true run-web-server=true ``` Point your web browser to `http://127.0.0.1:8080`, open the console and click on the producer ID, you should see such messages in the console once the video is displayed: ``` { "type": "InfoMessage", "mid": "video0", "info": { "meta": { "type": "timeCode", "hours": 0, "minutes": 0, "seconds": 5, "frames": 17, "field_count": 0, "fps": [ 30, 1 ], "flags": "none", "latest_daily_jam": "2024-08-23T17:51:42+0200" } } } ``` ## 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 ### WHIP Client WHIP Client Signaller uses BaseWebRTCSink Testing the whip client as the 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 ! \ whipclientsink 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. ### WHIP Server WHIP Server Signaller uses BaseWebRTCSrc The WHIP Server as the signaller can be tested in two ways. Note: The initial version of `whipserversrc` does not check any auth or encryption. Host application using `whipserversrc` behind an HTTP(s) proxy to enforce the auth and encryption between the WHIP client and server #### 1. Using the GStreamer element `whipclientsink` a. In one tab of the terminal start the WHIP server using the below command ``` shell RUST_BACKTRACE=full GST_DEBUG=webrtc*:6 GST_PLUGIN_PATH=target/x86_64-unknown-linux-gnu/debug:$GST_PLUGIN_PATH gst-launch-1.0 whipserversrc signaller::host-addr=http://127.0.0.1:8190 stun-server="stun://stun.l.google.com:19302" turn-servers="\<\"turns://user1:pass1@turn.serverone.com:7806\", \"turn://user2:pass2@turn.servertwo.com:7809\"\>" ! videoconvert ! autovideosink ``` b. In the second tab start the WHIP Client by sending a test video as shown in the below command ``` shell RUST_BACKTRACE=full GST_DEBUG=webrtc*:6 GST_PLUGIN_PATH=target/x86_64-unknown-linux-gnu/debug:$GST_PLUGIN_PATH gst-launch-1.0 videotestsrc ! videoconvert ! video/x-raw ! queue ! \ whipclientsink name=ws signaller::whip-endpoint="http://127.0.0.1:8190/whip/endpoint" ``` #### 2. Using Meetecho's `simple-whip-client` Set up the simple whip client using using the instructions present in https://github.com/meetecho/simple-whip-client#readme a. In one tab of the terminal start the WHIP server using the below command ``` shell RUST_BACKTRACE=full GST_DEBUG=webrtc*:6 GST_PLUGIN_PATH=target/x86_64-unknown-linux-gnu/debug:$GST_PLUGIN_PATH gst-launch-1.0 whipserversrc signaller::host-addr=http://127.0.0.1:8190 stun-server="stun://stun.l.google.com:19302" turn-servers="\<\"turns://user1:pass1@turn.serverone.com:7806\", \"turn://user2:pass2@turn.servertwo.com:7809\"\>" name=ws ! videoconvert ! autovideosink ws. ! audioconvert ! autoaudiosink ``` b. In the second tab start the `simple-whip-client` as shown in the below command ``` shell ./whip-client --url http://127.0.0.1:8190/whip/endpoint \ -A "audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc perfect-timestamp=true ! rtpopuspay pt=100 ssrc=1 ! queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload=100" \ -V "videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! vp8enc deadline=1 ! rtpvp8pay pt=96 ssrc=2 ! queue ! application/x-rtp,media=video,encoding-name=VP8,payload=96" \ -S stun://stun.l.google.com:19302 \ -l 7 \ -n true ``` Terminating the client will close the session and the client should receive 200 (OK) as the response to the DELETE request ## 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. ## Streaming from LiveKit using the livekitwebrtcsrc element First, publish a stream to the room using the following command: ```shell gst-launch-1.0 livekitwebrtcsink name=sink \ signaller::ws-url=ws://127.0.0.1:7880 \ signaller::api-key=devkey \ signaller::secret-key=secret \ signaller::room-name=testroom \ signaller::identity=gst-producer \ signaller::participant-name=gst-producer \ video-caps='video/x-h264' \ videotestsrc is-live=1 \ ! video/x-raw,width=640,height=360,framerate=15/1 \ ! timeoverlay ! videoconvert ! queue ! sink. ``` Then play back the published stream: ```shell gst-launch-1.0 livekitwebrtcsrc \ name=src \ signaller::ws-url=ws://127.0.0.1:7880 \ signaller::api-key=devkey \ signaller::secret-key=secret \ signaller::room-name=testroom \ signaller::identity=gst-consumer \ signaller::participant-name=gst-consumer \ signaller::producer-peer-id=gst-producer \ video-codecs='' \ src. ! queue ! videoconvert ! autovideosink ``` ### Auto-subscribe with livekitwebrtcsrc element With the LiveKit source element, you can also subscribe to all the peers in your room, simply by not specifying any value for `signaller::producer-peer-id`. Unwanted peers can also be ignored by supplying an array of peer IDs to `signaller::excluded-producer-peer-ids`. Importantly, it is also necessary to add sinks for all the streams in the room that the source element has subscribed to. First, publish a few streams using different connections: ```shell gst-launch-1.0 \ livekitwebrtcsink name=sinka \ signaller::ws-url=ws://127.0.0.1:7880 \ signaller::api-key=devkey \ signaller::secret-key=secret \ signaller::room-name=testroom \ signaller::identity=gst-producer-a \ signaller::participant-name=gst-producer-a \ video-caps='video/x-vp8' \ livekitwebrtcsink name=sinkb \ signaller::ws-url=ws://127.0.0.1:7880 \ signaller::api-key=devkey \ signaller::secret-key=secret \ signaller::room-name=testroom \ signaller::identity=gst-producer-b \ signaller::participant-name=gst-producer-b \ video-caps='video/x-vp8' \ livekitwebrtcsink name=sinkc \ signaller::ws-url=ws://127.0.0.1:7880 \ signaller::api-key=devkey \ signaller::secret-key=secret \ signaller::room-name=testroom \ signaller::identity=gst-producer-c \ signaller::participant-name=gst-producer-c \ video-caps='video/x-vp8' \ videotestsrc is-live=1 \ ! video/x-raw,width=640,height=360,framerate=15/1 \ ! timeoverlay ! videoconvert ! queue ! sinka. \ videotestsrc pattern=ball is-live=1 \ ! video/x-raw,width=320,height=180,framerate=15/1 \ ! timeoverlay ! videoconvert ! queue ! sinkb. videotestsrc is-live=1 \ ! video/x-raw,width=320,height=180,framerate=15/1 \ ! timeoverlay ! videoconvert ! queue ! sinkc. ``` Then watch only streams A and B by excluding peer C: ```shell gst-launch-1.0 livekitwebrtcsrc \ name=src \ signaller::ws-url=ws://127.0.0.1:7880 \ signaller::api-key=devkey \ signaller::secret-key=secret \ signaller::room-name=testroom \ signaller::identity=gst-consumer \ signaller::participant-name=gst-consumer \ signaller::excluded-producer-peer-ids='' \ src. ! queue ! videoconvert ! autovideosink src. ! queue ! videoconvert ! autovideosink ``` [LiveKit]: https://livekit.io/ [janus]: https://github.com/meetecho/janus-gateway [simple whip server]: https://github.com/meetecho/simple-whip-server/