WebRTC

What is WebRTC? #

WebRTC (Web Real-Time Communications) is a framework for real-time communication and in libp2p is used to establish browser-to-server and browser-to-browser connections between applications.

WebRTC was originally designed to make audio, video, and data communication between browsers user-friendly and easy to implement. It was first developed by Global IP Solutions (or GIPS). In 2011, GIPS was acquired by Google where the W3C started to work on a standard for WebRTC.

It serves as a good choice for applications that need built-in support for media communication and do not have specific requirements for the underlying transport protocol.

WebRTC in libp2p #

In libp2p, WebRTC is used as a transport protocol to connect from browsers to other nodes. However, libp2p does not make use of any of WebRTC’s multimedia features. The features employed in libp2p are:

Key features #

• Peer connections: WebRTC enables direct peer-to-peer connections between browsers and other nodes.

• Data channels: WebRTC provides peer-to-peer data channels, which works on SCTP (Stream Control Transmission Protocol) and use SDP (Session Description Protocol) to negotiate the parameters of the data channel.

  A WebRTC data channel allows applications to send a text or binary data over an active connection to a peer. This means libp2p can utilize data channels as a transport to send raw data to peers and enables applications to build anything they like.

• NAT traversal: WebRTC includes mechanisms (like ICE) to connect to nodes that run behind NATs and firewalls. In non-decentralized WebRTC, this can be facilitated by a TURN server., but other signaling channels, such as WebSocket running on a central server, can also be used. Using a custom signaling protocol or a different signaling service is also possible.

• Security: WebRTC connections are encrypted using DTLS. DTLS is similar to TLS but is designed to work on an unreliable transport instead of an ordered byte stream like TCP.

• API: Browsers expose an API to establish WebRTC connections. The RTCPeerConnection API allows two applications on different endpoints to communicate.

Browser-to-Server #

The first use case supported by a native WebRTC transport in libp2p is browser-to-server (as described in the specifications).

libp2p WebRTC enables browsers nodes to connect to public server nodes without those endpoints providing a TLS certificate within the browser’s trustchain.

In libp2p:

• WebRTC multiaddresses are composed of a standard UDP multiaddr, followed by webrtc and the multihash of the certificate that the node uses, as such: /ip4/1.2.3.4/udp/1234/webrtc/certhash/<hash>/p2p/<peer-id>;
• WebRTC encrypts connections using DTLS. However, an additional handshake is required to authenticate a peer’s peer ID once the WebRTC connection has been established.
• A browser can connect to a server node without needing a trusted TLS certificate.

Contrary to the standard WebRTC handshake process, the browser and server do not exchange the SDP Offer and Answer. Instead, they employ a technique known as SDP munging. This technique allows the browser node to simulate the exchange of an SDP, but in reality, it constructs it locally using the information provided by the server node’s multiaddress.

When establishing a WebRTC connection, the browser and server perform a standard DTLS handshake as part of the connection setup. Of the three primary focuses of information security, a successful DTLS handshake only provides two: confidentiality and integrity. Authenticity is achieved by succeeding the Noise handshake following the DTLS handshake.

Coming soon: Browser-to-Browser #

Eventually, libp2p will have support for communication between two browsers.

The technical specification and initial implementations of WebRTC Browser-to-Browser connectivity is planned for release in early 2023. Track the progress here.

Comparing WebRTC and WebTransport #

In general, WebRTC was primarily built for in-browser audio and video communication, whereas WebTransport aims to offer a general-purpose bidirectional byte-stream interface between a browser and a server.

Regarding connectivity, WebTransport only supports client-server connections, while WebRTC supports peer-to-peer connections. WebRTC is also more complex, as many underlying protocols are involved in creating a connection, as opposed to WebTransport, which only depends on QUIC.

Check out the WebTransport and WebRTC sections of the libp2p connectivity site to learn more.