Mastering WebRTC: A Step‑by‑Step Guide to Real‑Time Communication in the Browser

In the frontend field, WebRTC is a niche technology, but it is essential for online education. This article introduces the WebRTC workflow to give readers a complete understanding of the technology.

WebRTC (Web Real‑Time Communications)

WebRTC is an open‑source audio/video technology promoted by Google and standardized by the W3C, enabling real‑time peer‑to‑peer communication between browsers without intermediate media servers.

A typical WebRTC communication process consists of four steps: find the peer, negotiate, establish the connection, and start communication.

01 Find Peer

Before communication can start, both parties must know each other's existence, which requires a signaling server . The signaling server is a custom “middle‑man” that can be implemented with any technology (e.g., WebSocket or AJAX) to exchange session information such as room IDs and user IDs.

Both ends of a WebRTC session are called peers , and a successfully established connection is represented by a PeerConnection object. A single WebRTC session may contain multiple PeerConnection instances.

const pc2 = new RTCPeerConnection({ /* configuration */ });

02 Negotiation

The negotiation phase mainly involves SDP (Session Description Protocol) exchange . SDP describes media capabilities, ICE candidates, and transport protocols.

Media capabilities (codecs, etc.)

ICE candidate addresses

Transport protocols

SDP lines follow a simple type=value format. A typical SDP example:

v=0