gst-plugins-rs/net/webrtc/examples/README.md

# 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.

[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
```

### 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
```

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 -- --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 -- --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
```

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 -- --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 -- \
  --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 -- --expect-clock-signalling
```

#### Sender

```shell
cargo r --example webrtc-precise-sync-send -- --clock ntp --do-clock-signalling \
  --video-streams 0 --audio-streams 2
```