How WebRTC Works, Explained Simply (2026)

What is WebRTC?

WebRTC is a free toolkit built into major web browsers that enables live transmission of voice, video, and data between devices. For voice AI agents, WebRTC prioritizes speed over perfection. Unlike other web technologies that pause to wait for missing data packets, WebRTC skips lost pieces to maintain a natural conversation flow, as brief gaps are less disruptive than frozen pauses.

The Four Steps of a WebRTC Call

1. Swap Contact Cards (Signaling): The two devices exchange contact cards containing supported audio formats, potential connection addresses, and security codes. This exchange typically happens through a signaling server.
2. Find a Route: Since most devices sit behind firewalls or routers, they use a STUN server to discover their public addresses and establish a direct connection. If firewalls are too strict, a TURN server acts as a middleman to relay the audio.
3. Lock It (Encryption): All transmitted data is permanently encrypted. Security codes from the contact cards verify that the connection is secure and connected to the correct device. While a TURN relay cannot hear the encrypted audio, a media hub server can unless end-to-end encryption is implemented.
4. Talk: Audio is split into small pieces (typically 50 per second). The receiving device uses a jitter buffer to smooth out playback. Additionally, the browser performs echo cancellation and noise reduction to prevent the AI agent from hearing its own voice and interrupting itself.

AI Integration

The AI agent can connect directly to the browser or through a central hub called an SFU (Selective Forwarding Unit). Common frameworks and services utilizing WebRTC for voice AI include Daily (Pipecat), LiveKit, and OpenAI's Realtime API.