Overview

Session Initiation Protocol (SIP) interconnect refers to a configuration where two or more SIP-based networks or systems are linked to enable the exchange of voice traffic between them.

Stream SIP Interconnect enables VoIP calls from external third-party services to connect with Stream video/audio calls, creating a seamless bridge between participants joining through SIP networks and those connected via the Stream Client SDKs over WebRTC.

Currently, only inbound trunks are supported. This means external SIP calls can be routed into Stream calls, but outbound dialing from Stream to SIP endpoints is in development.

Features

  • Inbound Trunk — Configure SIP trunks with credentials and routing rules to receive calls on the Stream SIP bridge.
  • DTMF — Handle touch-tone (DTMF) input from SIP callers for IVR menus, pin-code entry, and more.
  • Dashboard & Debugging — Monitor active SIP calls, join calls for testing, and debug common issues from the Stream dashboard.

Supported Providers

Stream SIP Interconnect works with any standards-compliant SIP provider. We provide step-by-step guides for:

  • Twilio — Quick start using TwiML for SIP dial-in.
  • Telnyx — Integration using SIP Connections and Call Control.

Prerequisites

  • A Stream account with video/audio enabled.
  • A VoIP phone number purchased from a supported provider (Twilio, Telnyx, or another SIP-compliant service).
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