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Use SIP with Twilio Voice

Before you Begin

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Before you can use SIP Interface, you must sign up for a Twilio account (if you don't already have one). To sign up for an account click here(link takes you to an external page).

Connect your communications infrastructure to Twilio and start building programmable voice applications, such as call centers and IVRs, with Twilio's powerful and flexible voice capabilities. You can connect to Twilio over the public internet or alternatively via a private connection using Twilio's Interconnect(link takes you to an external page). Programmable Voice SIP lets you route your voice calls with global reach to any landline phone, mobile phone, browser, mobile app, or any other SIP endpoint.

The following diagram illustrates how the Twilio Cloud can sit in-between your existing infrastructure in call flows, allowing you to build programmable voice applications. Twilio is able to programmatically handle incoming calls from the PSTN and sessions from your SIP communications infrastructure such as a PBX.

Diagram showing how Twilio can sit in the middle of your existing communications infrastructure.

Session Initiation Protocol (SIP) is a standardized communications protocol that has been widely adopted for managing multimedia communication sessions for voice and video calls. SIP may be used to establish connectivity between your communications infrastructures such as an on-premise or virtual PBX and Twilio's communications platform.

Sending SIP to Twilio

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Twilio's Programmable Voice SIP Interface product enables you to use your existing SIP communications infrastructure to initiate SIP sessions with the Twilio Cloud. SIP Interface uses Twilio's TwiML language and/or Twilio's REST APIs to create advanced voice applications. Learn how to get started connecting your SIP communications infrastructure to the Twilio Cloud.

Receiving SIP from Twilio

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Twilio's Programmable Voice SIP Interface product enables your advanced voice applications to initiate SIP sessions from the Twilio Cloud towards your existing SIP communications infrastructure using Twilio's TwiML language and/or Twilio's REST APIs. Learn how to get started connecting the Twilio Cloud to your SIP communications infrastructure.

Make sure you are aware of the following Programmable Voice SIP Domain limits.

Twilio allows you to register your SIP Phones or SIP Endpoints with Twilio. SIP Registration is used to identify the location of the SIP Endpoints. Therefore, the user can receive calls irrespective of physical location of the SIP Endpoint.

This feature allows your SIP Endpoints can send REGISTER request to Twilio. For details see here.

Call Transfers using SIP REFER from Twilio

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Call transfer enables you to move an active call from one endpoint to another, in SIP this is accomplished using the SIP REFER method.

Twilio supports initiating SIP REFER method from Twilio towards your IP communications infrastructure leveraging the <Refer> verb.

SIP custom header allows you to send customized headers.

UUI (User-to-User Information) Header

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UUI header allows you to send contextual information over the SIP call. You can check Sending-sip with UUI and Receiving-sip with UUI for further UUI details.

Twilio supports RFC-2833(link takes you to an external page) for sending and receiving DTMF.

Twilio supports G.711 μ-law (PCMU) and A-law (PCMA)(link takes you to an external page) codecs for media. These are the most popular codecs used by carriers so transcoding is unnecessary.

Securing SIP Traffic using TLS

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Encryption ensures that the call signaling remains private during transmission. Transport Layer Security (TLS) provides encryption for SIP signaling.

In order for this to function properly, devices in your network that communicate directly with Twilio must be configured to trust Twilio's TLS/SSL Certificate. Twilio uses certificates issued by a CA (Certificate Authority). You may need to add additional root certificates to your communications infrastructure to establish the authenticity of Twilio's certificate on the network. Download Twilio's bundle of trusted CA certificates (last updated September 1, 2023).

Note: the current bundle contains the following root certificates:

  • DigiCert Global Root CA
  • DigiCert Global Root G2
  • DigiCert Global Root G3

Please be aware that the Twilio CA bundle may be updated in the future, for example when root certificates expire or are distrusted by the CA. In such cases we will notify you to update your SIP devices. Please ensure that your email address is up to date in your account to ensure you receive such communication.

There is no further configuration required for TLS and you can start sending over port 5061 straight away. TLS support via port 5061 is always active and does not require a manual toggle like Secure Media.

For calls where TLS is active this will cause SIP PCAPs downloaded from the console to be empty.

When sending TLS SIP traffic to Twilio, you will need to ensure that your infrastructure is using next-hop domain as opposed to next-hop ip. This is a common misconfiguration and will cause a 403 error that will not be visible on your Twilio account. Traffic must be sent to the domain {example}.sip.{edge}, rather than an IP address in order to associate the traffic with your Twilio account.

TLS Specifications

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  • Supported TLS versions: TLSv1.0 , TLSv1.1 and TLSv1.2 . PLEASE NOTE: To better comply with security requirements, we have deprecated TLSv1.0 and TLSv1.1 for inbound and outbound SIP calls, as well as SIP registration. If your SIP infrastructure requires using TLSv1.0 or TLSv1.1, you can configure your Twilio Account to allow these deprecated versions in your console under Voice → Settings → Allow Deprecated SIP/TLS versions. If this setting is enabled, your SIP endpoints can use the deprecated TLSv1.0 and TLSv1.1 versions for SIP signaling sent to or received from Twilio. If disabled, only non-deprecated TLSv1.2+ is allowed. Twilio strongly recommends the use of TLS version 1.2 when connecting your SIP infrastructure.


Note: Twilio SIP Interface outbound call URI configurations using the sips URI scheme in order to enable end-to-end encryption is NOT supported by Twilio. However, we do support sip URI schemes using transport=tls for point-to-point encryption.

If you configure your SIP Interface URIs to use sips schemes, these sips URIs will be handled as if they were sip URIs using TLS transport. Twilio will effectively adjust the URI internally to instead be routed using the sip scheme and transport=tls on the outbound messages, resulting in point-to-point encryption between Twilio and the customer equipment.

Twilio strongly suggests not using sips schemes in your Twilio SIP configurations, as this could cause possibly unintended behavior, due to how we process such URIs. Instead, we suggest using sip schemes with TLS transport. This method, along with the security of our voice architecture and Super Network, is an effective way of adding encryption to your Twilio SIP connections.

Secure Media uses encryption to ensure that the call media and associated signaling remains private during transmission. Secure Real-Time Protocol (SRTP) provides encryption for media. For details see here.

Prepare your communications infrastructure to make sure that your SIP infrastructure has connectivity to the Twilio Cloud and vice versa. To ensure that your communications infrastructure doesn't block communication, you must update your list of allowed IP addresses. We strongly encourage you to allow all of the following IP address ranges and ports on your firewall for SIP signaling and RTP media traffic.

This is important if you have Numbers in different regions as well as for availability purposes (e.g. if North America Virginia gateways are down, then North America Oregon gateways will be used).

Please see Twilio's SIP IP addresses for the complete list.

Communications Infrastructure

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A broad term to refer to IP-PBX, SBC, IP-phones, etc…

IP-phone or a soft client with which a user initiates a VoIP call

Equivalent to a SIP phone number and takes the form, sip:username@SIPDomain

It takes the form {example}.sip.{region} where {example} is specified by the customer and {region}is the data center where the registrar is located. Initially only us1.

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