# WebRTC

**Source:** https://promtable.com/glossary/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.

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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

- 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.

## Related terms

- [voice-pipeline](https://promtable.com/glossary/voice-pipeline)
- [realtime-api](https://promtable.com/glossary/realtime-api)

## Sources

- [WebRTC overview](https://webrtc.org/)

*Last updated: 2026-06-01*
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Original page: https://promtable.com/glossary/webrtc
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