WebRTC
WebRTC (Web Real-Time Communication) is the browser-native standard for low-latency audio + video + data peer connections — the foundation for browser-based voice agents, video calls (Zoom Web, Meet, ChatGPT Voice), and live streaming in 2026.
WebRTC ships in every modern browser + mobile WebView: signaling protocol (SDP offer / answer over WebSocket), media transport (RTP + DTLS-SRTP encrypted UDP), NAT traversal (STUN / TURN servers), echo cancellation, jitter buffering, automatic codec negotiation (Opus for audio, VP8 / VP9 / AV1 for video). For voice agents, WebRTC provides sub-200ms audio transport (vs HTTP / WebSocket's higher latency + jitter). Production deployments require TURN servers (for NAT-restricted clients), media servers (LiveKit, Janus, mediasoup) for scale, and SFU (Selective Forwarding Unit) architectures for multi-party calls. By 2026 WebRTC is the de-facto standard for production voice agents in browser-native contexts; native mobile apps often still use platform-specific networking.
When to use webrtc
- Browser-based voice / video agents.
- Low-latency real-time apps.
Common mistakes
- Skipping TURN — clients behind strict NAT can't connect.
- P2P at scale — N×N connections collapse beyond ~6 peers; use SFU.
FAQ
What is webrtc?
WebRTC (Web Real-Time Communication) is the browser-native standard for low-latency audio + video + data peer connections — the foundation for browser-based voice agents, video calls (Zoom Web, Meet, ChatGPT Voice), and live streaming in 2026.
When should I use webrtc?
Browser-based voice / video agents. Low-latency real-time apps.
What are the most common mistakes with webrtc?
Skipping TURN — clients behind strict NAT can't connect. P2P at scale — N×N connections collapse beyond ~6 peers; use SFU.
Related terms
- Voice pipeline — A voice pipeline is the chain of audio processing stages — VAD → STT → LLM → TTS → playback — composed in a streaming framework like Pipecat, LiveKit Agents, Vocode, or a managed platform's internal stack.
- Realtime API — A Realtime API is the WebSocket / WebRTC-based LLM endpoint that supports streaming audio in + audio out for natural duplex conversation — OpenAI Realtime API, Gemini Live, ElevenLabs Conversational, Cartesia Sonic are 2026 leaders.
Sources
Last updated: 2026-06-01. Raw markdown: https://promtable.com/glossary/webrtc.md.