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.

Note:

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).