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The WebRTC Gateway enables real-time voice communication between a web browser and a PolyAI voice agent, using WebSocket signaling and bidirectional WebRTC audio. It provides two integration layers:
  • WebSocket signaling for session setup, SDP exchange, and ICE candidate exchange
  • WebRTC media for bidirectional audio once the connection is established

Prerequisites

  • A WebRTC-capable browser
  • Microphone permissions enabled
  • A PolyAI authentication token

Quick start

  1. Open a WebSocket connection to the signaling endpoint
  2. Create a WebRTC peer connection and collect microphone audio
  3. Send an offer message containing SDP and your auth token
  4. Receive an answer message containing SDP and a session identifier
  5. Exchange ICE candidates until the connection is established
  6. Audio flows bidirectionally

Authentication token

The authToken you send in the offer message is the same credential used to authenticate SIP traffic — your connector token (the value provided by PolyAI for the X-PolyAi-Auth-Token SIP header). The gateway accepts two token types: For your own WebRTC client (widgets, native apps, custom browser integrations), use the connector token.

Signaling endpoint

Signaling URL (WebSocket): wss://webrtc-gateway.us-1.platform.polyai.app/api/v1/webrtc/signal All signaling messages are JSON objects sent over the WebSocket connection.

Message structure

All signaling messages follow the same top-level structure.

Message types

Offer (client to server)

Starts a new session. Send with an empty sessionId. Example message:

Answer (server to client)

Sent in response to an offer. Contains the SDP answer and the assigned sessionId.
Store the sessionId and use it for all subsequent messages.

ICE candidate (bidirectional)

Sent by both client and server to establish network connectivity.

Close (client to server)

Terminates the session gracefully.

Error (server to client)

Sent when an error occurs.
Error codes:

WebRTC configuration

Audio codec

The gateway requires Opus audio.
  • MIME type: audio/opus
  • Sample rate: 48 kHz
  • Channels: stereo

ICE servers

Configure your peer connection with a STUN server. TURN is recommended for restrictive networks. Example STUN server: stun.l.google.com:19302

Browser support

  • Chrome 72 or newer
  • Firefox 60 or newer
  • Safari 14.1 or newer
  • Edge 79 or newer

Troubleshooting

Unauthorized error

Ensure the authentication token is valid and included in the offer message. Every offer requires an authToken, including studio draft and preview calls that pin a specific build with agentVersionOverride. Offers without a token are rejected with an UNAUTHORIZED error and the message Auth token required.

No audio

  • Confirm microphone permissions are granted
  • Verify Opus is negotiated successfully

ICE connection fails

  • Corporate firewalls may require TURN
  • Ensure UDP traffic is allowed
  • Configure TURN over TCP if needed
Last modified on July 13, 2026