GStreamer WebRTC demos

All demos use the same signalling server in the signalling/ directory

Filing bugs

Please only file bugs about the demos here. Bugs about GStreamer's WebRTC implementation should be filed on the GStreamer bugzilla.

You can also find us on IRC by joining #gstreamer @ FreeNode.

Documentation

Currently, the best way to understand the API is to read the examples. This post breaking down the API should help with that:

http://blog.nirbheek.in/2018/02/gstreamer-webrtc.html

sendrecv: Send and receive audio and video

• Serve the js/ directory on the root of your website, or open https://webrtc.nirbheek.in

  • The JS code assumes the signalling server is on port 8443 of the same server serving the HTML

• Open the website in a browser and ensure that the status is "Registered with server, waiting for call", and note the id too.

Running the Rust version

• Install a recent Rust toolchain, e.g. via rustup.
• Run cargo build for building the executable.
• Run cargo run -- --peer-id=ID with the id from the browser. You will see state changes and an SDP exchange.

You can pass a --server argument to all versions, for example --server=wss://127.0.0.1:8443.