WebRTC (Web Real-Time Communication) was announced by Google in May 2011 and became a W3C and IETF standard over the following decade. It enables audio, video, and data sharing directly between browsers without requiring plugins, downloads, or third-party software. By 2023, WebRTC was supported by all major browsers and used by billions of people — most of whom have no idea what it is, because it operates invisibly in Google Meet, WhatsApp Web, Discord, and voice-note-enabled chat applications.
The Three Core Components
WebRTC has three primary APIs. getUserMedia accesses the device's camera and microphone with user permission — the browser's native capability to capture audio and video. RTCPeerConnection manages the peer-to-peer connection between two browsers, including codec negotiation, network traversal, and encryption. RTCDataChannel provides a bidirectional data channel between peers that can carry arbitrary data — text, files, or custom protocol messages — with latency characteristics suitable for real-time applications.
The ICE/STUN/TURN Infrastructure
Establishing a direct peer-to-peer connection between two browsers is complicated by NAT (Network Address Translation) — the mechanism that allows many devices to share a single public IP address. STUN (Session Traversal Utilities for NAT) servers help browsers discover their public IP addresses and negotiate direct connections when possible. When direct connection is impossible (behind symmetric NATs or firewalls), TURN (Traversal Using Relays around NAT) servers relay media between peers. About 85% of WebRTC connections can be established directly (via STUN); the remaining 15% require relay via TURN servers.
WebRTC Security
WebRTC mandates encryption: all audio, video, and data channels are encrypted using DTLS (Datagram Transport Layer Security) and SRTP (Secure Real-time Transport Protocol). This encryption is built into the protocol specification rather than being optional — unlike some earlier VoIP technologies. For anonymous chat, this means voice notes and audio exchanged via WebRTC are encrypted in transit without any additional configuration. The challenge for anonymous platforms is that STUN/TURN server infrastructure can log connection metadata (which IPs are connecting to which), creating a potential metadata exposure that careful infrastructure operators minimize through short-retention log policies.