SIP Interface over Twilio Interconnect

To connect over Twilio Interconnect, point your communications infrastructure to the following localized SIP Domain URIs:

In order for your SIP Endpoint to receive calls from Twilio, you will use the same TwiML or REST API calls you use today. The only difference is that you will now be specifying your Twilio Interconnect Connection, by including the edge parameter in the URI with the value of the Twilio Interconnect Edge Location where your private connection is configured, for example:

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    <Response>
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        <Dial>
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            <Sip>
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                sip:yourusername@example.com;edge={EDGE_LOCATION}
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            </Sip>
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        </Dial>
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    </Response>
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You may also use the deprecated tnx parameter in the URI with the SID value of the desired Twilio Interconnect connection, however, it is preferred that you use the edge parameter as documented above. An example of using the tnx parameter:

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    <Response>
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        <Dial>
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            <Sip>
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                sip:yourusername@example.com;tnx={TNX_SID}
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            </Sip>
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        </Dial>
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    </Response>
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Interconnect Connections - Global Media IP Range

The Interconnect Connections Destination IP Ranges and Port Ranges are now identical across all locations:

† The SDK will select any available port from the ephemeral range. On most machines, this means the port range 1,024 to 65,535.

We strongly encourage you to allow all of Twilio's following IP address ranges and ports on your firewall for SIP signalling traffic. This is important if you have Numbers in different regions and for resilience purposes (e.g. if North America Virginia gateways are down, then North America Oregon gateways will be used).

This guide teaches the basics required for the following use cases:

You can use this guide to securely route incoming calls from the PSTN to your private SIP infrastructure for inbound contact center agents.

To learn more advanced features that you can use with inbound contact centers, see Voice inbound call center.

You can use this guide to send SIP traffic from your private infrastructure to Twilio's SIP Interface to place outbound calls to the PSTN.

To learn more advanced features that you can use with outbound call centers, see Voice outbound call center.

You can use this guide to bridge your private SIP network with the public switched telephone network (PSTN) using a dedicated, private connection.

To learn more advanced features that you can use with PSTN connectivity, see Voice PSTN connectivity.

After following this guide, you have established a private, secure path for SIP signaling and media between your infrastructure and Twilio. You can confirm it worked by placing a test call and verifying that the traffic traverses your Interconnect connection rather than the public internet, and by checking that signaling reaches the correct localized SIP URI.

Explore the following guides to build on what you've learned in this guide:

• Emergency calling for SIP interfaces: Configure emergency call routing and address validation for your SIP infrastructure.
• SIP: Explore the foundational concepts and configuration options for Twilio SIP Interfaces.
• How to add programmability to your existing SIP network: Learn how to wrap your SIP calls in TwiML to add features like recording and IVRs.
• SIP API: Manage your SIP Domains, IP access control lists, and credentials programmatically.
• Secure media: Implement SRTP to encrypt your media streams for enhanced security.

Find reference documentation for the topics on this page.

• SIP Domain resource
• SIP Credential resource
• SIP IpAddress resource
• TwiML™ Voice: <Sip>
• Understanding Twilio media edges