Web Real-Time Communications (WebRTC)
Video calls (Meet, Zoom web), voice calls, peer-to-peer file transfer, and collaborative tools. The browser API surface for real-time A/V communication.
What It Defines
W3C API enabling direct peer-to-peer audio, video, and data communication between browsers without a server relay. Relies on ICE for NAT traversal, DTLS for encryption, and SRTP for media encryption.
Canonical (Normative)
Convenient (Practical)
Publishes web platform specs including CSS, accessibility, security policies, Service Workers, Web App Manifest, and many browser APIs. Also maintains some versioned HTML/DOM specs.
Related Specs
NAT traversal is why WebRTC calls fail. ICE is how browsers find a working path between peers. Know it when debugging call connectivity failures.
STUN is how WebRTC discovers the public-facing address for peer-to-peer connections. Every WebRTC deployment needs a STUN server.
~15-20% of WebRTC calls require TURN relay because both peers are behind symmetric NAT. Without TURN, those calls fail silently.
WebRTC media is always encrypted. DTLS-SRTP is the protocol that does it. Know it when troubleshooting media security negotiation or DTLS handshake failures.