init webrtc content

content/concepts/transports/webrtc.md

@@ -1,9 +1,110 @@
 ---
 title: "WebRTC"
-description: "WebRTC is a new specification that uses QUIC to offer an alternative to WebSocket. Conceptually, it can be considered WebSocket over QUIC.Learn about WebTransport and how it is used in libp2p."
+description: "WebRTC is a protocol for real-time communication and is used to establish connections between browsers and other applications."
 weight: 110
 aliases:
     - "/concepts/transports/webrtc"
 ---
 
-Coming soon!
+## What is WebRTC?
+
+WebRTC (Web Real-Time Communications) is a framework for real-time communication
+and is used to establish connections between browsers and other applications.
+WebRTC 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.
+
+### Key features
+
+- Audio and video support: WebRTC provides built-in support via a media stream API
+  that controls the multimedia activities of a device over the data consumed. This
+  allows applications to incorporate real-time audio and video streams easily.
+
+- Peer-to-peer communication: WebRTC enables direct communication between browsers
+  and other nodes without needing an intermediate server. This allows for faster
+  and more efficient communication. Peers can also retrieve or consume the media and
+  also produce it.
+
+- Data channel: WebRTC provides a data channel that allows applications to transfer
+  arbitrary data between peers. This works on
+  [SCTP (Stream Control Transmission Protocol)](https://en.wikipedia.org/wiki/Stream_Control_Transmission_Protocol) and reduces network congestion over UDP.
+
+- [NAT traversal](../nat/overview): WebRTC includes mechanisms (like
+  [ICE](https://datatracker.ietf.org/doc/rfc5245/))to connect to nodes that live behind
+  NATs and firewalls.
+
+- [Security](../secure-comm/overview): WebRTC includes built-in security features, and
+  connections are always encrypted by using [DTLS](https://en.wikipedia.org/wiki/Datagram_Transport_Layer_Security) or [SRTP](https://en.wikipedia.org/wiki/Secure_Real-time_Transport_Protocol).
+
+WebRTC includes several APIs to help facilitate the creation of a secure connection
+over the web.
+
+The `RTCPeerConnection` API allows two applications on different
+endpoints to communicate using a peer-to-peer protocol. The `PeerConnection` API
+interacts closely with a `getUserMedia` API for accessing a node's media-based peripheral
+device and uses the `getDisplayMedia` API to capture screen content. WebRTC allows a node
+to send and receive streams that include media content and arbitrary binary data
+through the `DataChannel`.
+
+### WebRTC and WebTransport
+
+While WebRTC and WebTransport are both web-based approaches that enable real-time
+communication between nodes, there are key differences. WebRTC supports peer-to-peer
+connections, while WebTransport only supports client-server connections.
+
+The underlying protocols in WebRTC and WebTransport are different, although both
+protocols share many of the same properties. WebTransport is also an
+alternative to the data channels available in WebRTC.
+
+WebRTC is also more complex, as there are many underlying protocols involved in order
+to create an active transport, as opposed to WebTransport, that uses QUIC.
+
+When connecting to a WebSocket server or when using plain TCP or QUIC connections,
+browsers require the server to present a TLS certificate signed by a trusted CA
+(certificate authority). Few nodes have such a certificate. One method to overcome
+this is to use the [WebTransport](webtransport) browser API that offers a way to
+accept a server's certificate by checking the (SHA-256) hash of the certificate.
+
+However, a certificate is still needed, even if it is "just" self-signed.
+The browser must also know the certificate hash. WebRTC can overcome this and
+does not require a trusted certificate.
+> While WebRTC does not require the use of trusted certificates, it does not
+> eliminate their usage, as  WebRTC relies on TLS to establish secure connections
+> between peers and to protect the data being transferred.
+
+## WebRTC in libp2p
+
+Libp2p WebRTC enables browsers to connect to public server nodes (and eventually,
+browsers to connect to other browsers) without those endpoints providing a TLS
+certificate within the browser's trustchain.
+
+In libp2p:
+
+- the `RTCPeerConnection` API allows an application to establish peer-to-peer
+  communications;
+- the `RTCDataChannel` API supports peer-to-peer data channels;
+- a 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>`.
+- The TLS certificate fingerprint in `/certhash` is a multibase encoded multihash.
+- WebRTC can support UDP and TCP, but implementations must always support UDP.
+
+### Browser-to-Server
+
+A browser can connect to a server node without needing a trusted TLS
+certificate. View [this example](https://github.com/libp2p/specs/blob/master/webrtc/README.md#browser-to-public-server) on browser-to-server connection establishment in the technical
+specification for reference.
+
+<!-- TO ADD: DIAGRAMS ONCE READY + CONTEXt -->
+
+### Coming soon: Browser-to-Browser
+
+Eventually, libp2p will have support for two-way communication between two
+browsers in real-time.
+
+The technical specification and initial implementations of WebRTC
+Browser-to-Browser connectivity is planned for release in early 2023.
+Track the progress [here](https://github.com/libp2p/specs/issues/475).
+
+{{< alert icon="💡" context="note" text="See the WebRTC <a class=\"text-muted\" href=\"https://github.com/libp2p/specs/blob/master/webrtc/README.md\">technical specification</a> for more details." />}}