Skip to main content
The WebRTC Gateway enables real-time voice communication between a web browser and a PolyAI voice agent. It has two 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

​
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.
FieldRequiredDescription
typeYesMessage type: offer, answer, ice-candidate, error, close
sessionIdYesEmpty string when creating a new session
dataNoMessage-specific payload
authTokenOffer onlyAuthentication token
callSidNoOptional call identifier
callerNoOptional caller identifier
calleeNoOptional callee identifier

​
Message types

​
Offer (client to server)

Starts a new session. Send with an empty sessionId. Example message: 
{
  "type": "offer",
  "sessionId": "",
  "data": {
    "type": "offer",
    "sdp": "v=0 o=- 4611731400430051336 2 IN IP4 127.0.0.1"
  },
  "authToken": "your-auth-token",
  "callSid": "call-unique-id",
  "caller": "+14155551234",
  "callee": "+14155555678"
}

​
Answer (server to client)

Sent in response to an offer. Contains the SDP answer and the assigned sessionId. 
{
  "type": "answer",
  "sessionId": "550e8400-e29b-41d4-a716-446655440000",
  "data": {
    "type": "answer",
    "sdp": "v=0 o=- 4611731400430051336 2 IN IP4 192.168.1.1"
  }
}
Store the sessionId and use it for all subsequent messages.

​
ICE candidate (bidirectional)

Sent by both client and server to establish network connectivity. 
{
  "type": "ice-candidate",
  "sessionId": "550e8400-e29b-41d4-a716-446655440000",
  "data": {
    "candidate": "candidate:1 1 UDP 2130706431 192.168.1.1 54321 typ host",
    "sdpMid": "0",
    "sdpMLineIndex": 0
  }
}

​
Close (client to server)

Terminates the session gracefully. 
{
  "type": "close",
  "sessionId": "550e8400-e29b-41d4-a716-446655440000"
}

​
Error (server to client)

Sent when an error occurs. 
{
  "type": "error",
  "sessionId": "550e8400-e29b-41d4-a716-446655440000",
  "data": {
    "code": "UNAUTHORIZED",
    "message": "Invalid authentication token"
  }
}

​
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.

​
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