# webrtcsink examples Collection of webrtcsink examples ## webrtcsink-stats-server A simple application that instantiates a webrtcsink and serves stats over websockets. The application expects a signalling server to be running at `ws://localhost:8443`, similar to the usage example in the main README. ``` shell cargo run --example webrtcsink-stats-server ``` Once it is running, follow the instruction in the webrtcsink-stats folder to run an example client. ## webrtcsink-custom-signaller An example of custom signaller implementation, see the corresponding [README](webrtcsink-custom-signaller/README.md) for more details on code and usage. ## WebRTC precise synchronization example This example demonstrates a sender / receiver setup which ensures precise synchronization of multiple streams in a single session. [RFC 6051]-style rapid synchronization of RTP streams is available as an option. Se the [Instantaneous RTP synchronization...] blog post for details about this mode and an example based on RTSP instead of WebRTC. The examples can also be used for [RFC 7273] NTP or PTP clock signalling and synchronization. Finally, raw payloads (e.g. L24 audio) can be negotiated. Note: you can have your host act as an NTP server, which can help the examples with clock synchronization. For `chrony`, this can be configure by editing `/etc/chrony.conf` and uncommenting / editing the `allow` entry. The examples can then be launched with `--ntp-server _ip_address_`. [RFC 6051]: https://datatracker.ietf.org/doc/html/rfc6051 [RFC 7273]: https://datatracker.ietf.org/doc/html/rfc7273 [Instantaneous RTP synchronization...]: https://coaxion.net/blog/2022/05/instantaneous-rtp-synchronization-retrieval-of-absolute-sender-clock-times-with-gstreamer/ ### Signaller The example uses the default WebRTC signaller. Launch it using the following command: ```shell cargo run --bin gst-webrtc-signalling-server --no-default-features ``` ### Receiver The receiver awaits for new audio & video stream publishers and render the streams using auto sink elements. Launch it using the following command: ```shell cargo r --example webrtc-precise-sync-recv --no-default-features ``` The default configuration should work for a local test. For a multi-host setup, see the available options: ```shell cargo r --example webrtc-precise-sync-recv --no-default-features -- --help ``` E.g.: the following will force `avdec_h264` over hardware decoders, activate debug logs for the receiver and connect to the signalling server at the specified address: ```shell GST_PLUGIN_FEATURE_RANK=avdec_h264:MAX \ WEBRTC_PRECISE_SYNC_RECV_LOG=debug \ cargo r --example webrtc-precise-sync-recv --no-default-features -- \ --server 192.168.1.22 ``` ### Sender The sender publishes audio & video test streams. Launch it using the following command: ```shell cargo r --example webrtc-precise-sync-send --no-default-features ``` The default configuration should work for a local test. For a multi-host setup, to set the number of audio / video streams, to enable rapid synchronization or to force the video encoder, see the available options: ```shell cargo r --example webrtc-precise-sync-send --no-default-features -- --help ``` E.g.: the following will force H264 and `x264enc` over hardware encoders, activate debug logs for the sender and connect to the signalling server at the specified address: ```shell GST_PLUGIN_FEATURE_RANK=264enc:MAX \ WEBRTC_PRECISE_SYNC_SEND_LOG=debug \ cargo r --example webrtc-precise-sync-send --no-default-features -- \ --server 192.168.1.22 --video-caps video/x-h264 ``` ### The pipeline latency The `--pipeline-latency` argument configures a static latency of 1s by default. This needs to be higher than the sum of the sender latency and the receiver latency of the receiver with the highest latency. As this can't be known automatically and depends on many factors, this has to be known for the overall system and configured accordingly. The default configuration is on the safe side and favors synchronization over low latency. Depending on the use case, shorter or larger values should be used. ### RFC 7273 NTP or PTP clock signalling and synchronization For [RFC 7273] NTP or PTP clock signalling and synchronization, you can use commands such as: #### Receiver ```shell cargo r --example webrtc-precise-sync-recv --no-default-features -- \ --expect-clock-signalling ``` #### Sender ```shell cargo r --example webrtc-precise-sync-send --no-default-features -- \ --clock ntp --do-clock-signalling \ --video-streams 0 --audio-streams 2 ``` ### Raw payload The sender can be instructed to send raw payloads. Note that raw payloads are not activated by default and must be selected explicitly. This command will stream two stereo L24 streams: ```shell cargo r --example webrtc-precise-sync-send --no-default-features -- \ --video-streams 0 \ --audio-streams 2 --audio-codecs L24 ``` Launch the receiver with: ```shell cargo r --example webrtc-precise-sync-recv --no-default-features -- \ --audio-codecs L24 ``` This can be used to stream multiple RAW video streams using specific CAPS for the streams and allowing fallback to VP8 & OPUS if remote doesn't support raw payloads: ```shell cargo r --example webrtc-precise-sync-send --no-default-features -- \ --video-streams 2 --audio-streams 1 \ --video-codecs RAW --video-codecs VP8 --video-caps video/x-raw,format=I420,width=400 \ --audio-codecs L24 --audio-codecs OPUS --audio-caps audio/x-raw,rate=48000,channels=2 ``` ## Android ### `webrtcsrc` based Android application An Android demonstration application which retrieves available producers from the signaller and renders audio and video streams. **Important**: in order to ease testing, this demonstration application enables unencrypted network communication. See `app/src/main/AndroidManifest.xml` for details. #### Build the application * Download the latest Android prebuilt binaries from: https://gstreamer.freedesktop.org/download/ * Uncompress / untar the package, e.g. under `/opt/android/`. * Define the `GSTREAMER_ROOT_ANDROID` environment variable with the directory chosen at previous step. * Install a recent version of Android Studio (tested with 2023.3.1.18). * Open the project from the folder `android/webrtcsrc`. * Have Android Studio download and install the required SDK & NDK. * Click the build button or build and run on the target device. * The resulting `apk` is generated under: `android/webrtcsrc/app/build/outputs/apk/debug`. For more details, refer to: * https://gstreamer.freedesktop.org/documentation/installing/for-android-development.html Once the SDK & NDK are installed, you can use `gradlew` to build and install the apk (make sure the device is visible from adb): ```shell # From the android/webrtcsrc directory ./gradlew installDebug ``` #### Install the application Prerequisites: activate developer mode on the target device. There are several ways to install the application: * The easiest is to click the run button in Android Studio. * You can also install the `apk` using `adb`. Depending on your host OS, you might need to define `udev` rules. See: https://github.com/M0Rf30/android-udev-rules #### Setup 1. Run the Signaller from the `gst-plugins-rs` root directory: ```shell cargo run --bin gst-webrtc-signalling-server ``` 2. In the Android app, tap the 3 dots button -> Settings and edit the Signaller URI. 3. Add a producer, e.g. using `gst-launch` & `webrtcsink` or run: ```shell cargo r --example webrtc-precise-sync-send ``` 4. Click the `Refresh` button on the Producer List view of the app.