SIP Routing

Inbound SIP routing uses the destination URI from the INVITE. The gateway extracts the destination user and matches it against dynamic WebSocket routes first, then static HTTP routes.

Route Types

Exact match:

{ "match": "exact", "value": "support" }

Prefix match:

{ "match": "userPrefix", "value": "tenant-" }

Dynamic Routes

Dynamic routes are registered over the control WebSocket with route.register. They are owned by a single connection and removed when the connection closes.

Dynamic routes are the right default when an application needs:

• Pre-media SIP answer/reject.
• Session ownership.
• Cleanup when the backend disconnects.
• Events delivered on the same control channel.

Static HTTP Routes

Static routes are loaded from ROUTES_JSON and post INVITE payloads to a configured URL. They are useful for simple webhook integrations.

Outbound SIP

Applications create outbound SIP calls with sip.createOutbound on the control WebSocket or POST /calls over HTTP. The application supplies the request URI and local SDP offer. The gateway creates a Drachtio UAC dialog and returns the remote SDP answer.

For carriers that challenge outbound requests with SIP digest authentication, include both a username and password:

{
  "requestUri": "sip:15551234567@carrier.example.com;transport=tcp",
  "sdp": "...",
  "auth": {
    "username": "carrier-user",
    "password": "carrier-password"
  }
}

Follow-Up Events

Answered, re-INVITE, BYE, and terminated events are delivered to the owning WebSocket connection or to the HTTP receiver URL selected during the INVITE response.