WebRTC bundles several IETF specs into one cohesive browser API. The transport is UDP with DTLS-SRTP for encryption; connectivity uses ICE with STUN and TURN; media is RTP and feedback is RTCP; signalling (SDP offer/answer) is done by the application over its own channel (typically WebSocket). The API surface — RTCPeerConnection, getUserMedia, RTCDataChannel — has been a W3C Recommendation since January 2021 (W3C "WebRTC 1.0").
In streaming, WebRTC has two distinct roles. Peer-to-peer (the original use case) connects two browsers for video calls — Google Meet, Zoom Web, Whereby. Egress (server-as-source) uses WebRTC to deliver one stream to many viewers via an SFU — Twitch's low-latency mode, Kick, sports betting platforms. Both roles share the same protocol stack but have different scaling characteristics.
WebRTC's strength is sub-second latency and ubiquitous browser support. Its weakness is operational complexity — STUN/TURN deployment, SFU scaling, congestion control tuning. The 2022–2026 trend is to wrap WebRTC behind simpler protocols: WHIP for ingest, WHEP for delivery, Media over QUIC as a longer-term successor. Cloud providers (Cloudflare Calls, Mux, Dolby.io, LiveKit Cloud, AWS) sell managed WebRTC to hide the complexity. Self-hosted WebRTC remains viable but demands real expertise.
The book · Volume 3 of 7
Video Streaming: A Complete Guide to Video Delivery: From ABR and CDNs to Players, WebRTC, DRM, and the Economics of Streaming
WebRTC gives browsers sub-second audio and video, but connectivity, bandwidth estimation and server topology are where real products struggle. This volume covers WebRTC in detail, with SFU and MCU topologies, NAT traversal with relays, simulcast and SVC, congestion control in real-time calls and the economics of running media servers. Written by Nikolay Sapunov, CEO at Fora Soft.
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