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Use SIP with Twilio Voice

Before you Begin

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.

Overview

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. 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 the position of the Twilio Cloud in the call flows.

Programmable Voice SIP Diagram

What is SIP?

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

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

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.

Limits

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

Features

SIP Registration

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

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.

Call Transfers using SIP REFER to Twilio

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's programmable SIP call supports "blind" call transfers. This means you're now able to request a call be transferred by sending Twilio a SIP REFER message from your SIP communications infrastructure. See details here.

Parallel and Serial call using SIP

This feature allows customers to dial multiple SIP endpoints both sequentially and parallelly. See details here.

SIP Custom Header

SIP custom header allows you to send customized headers.

UUI (User-to-User Information) Header

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.

DTMF

Twilio supports RFC-2833 for sending and receiving DTMF.

Media codec

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

Securing SIP Traffic using TLS

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

Secure Media

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.

TLS/SRTP Specifications

  • SIP TLS
    • Versions: Twilio supports TLSv1.0, TLSv1.1 and TLSv1.2.
      Note: Twilio strongly recommends the use of TLS version 1.2.
    • Ciphers: ECDHE-ECDSA-AES128-GCM-SHA256, ECDHE-RSA-AES128-GCM-SHA256, ECDHE-ECDSA-AES128-SHA256, ECDHE-RSA-AES128-SHA256, ECDHE-ECDSA-AES256-GCM-SHA384, ECDHE-RSA-AES256-GCM-SHA384, ECDHE-ECDSA-AES256-SHA384, ECDHE-RSA-AES256-SHA384, AES128-GCM-SHA256, AES128-SHA256, AES128-SHA, AES256-GCM-SHA384, AES256-SHA256, AES256-SHA
    • If you are using TwiML to send SIP from Twilio, to enable encryption you must use the transport=tls parameter in your SIP noun in your Dial verb.
    • By default port 5061 will be used for TLS, however, you may specify the port you wish to use in your URI.
  • Secure Media
    • Sending SRTP to Twilio: Twilio supports the following Crypto suites: AES_CM_128_HMAC_SHA1_80 and AES_CM_128_HMAC_SHA1_32. Both may be included in your order of preference.
    • Receiving SRTP from Twilio: Only a single crypto suite will be advertised: AES_CM_128_HMAC_SHA1_80
  • Importing Twilio's Root CA Certificate TLS is used to encrypt SIP signaling between SIP endpoints. In order for this to function properly it is required that certain devices in the network import a CA certificate. Twilio uses certificates from a CA (Certificate Authority). It is important that you add the following root certificate to your communications infrastructure to establish its authenticity. Download Twilio's CA certificate.

    It is important to note that Twilio uses a wildcard certificate which can be used for multiple subdomains of a domain (*.sip.twilio.com). If your network element does not support wildcarded certificates please disable certificate validation.

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.

IP addresses

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).

North America Virginia Gateways:

Signalling IPs:

54.172.60.0
54.172.60.1
54.172.60.2
54.172.60.3
Ports: 5060 (UDP/TCP), 5061 (TLS)

Media IPs:

54.172.60.0/23
34.203.250.0/23
Port Range: 10,000 to 20,000 (UDP)

North America Oregon Gateways:

Signalling IPs:

54.244.51.0
54.244.51.1
54.244.51.2
54.244.51.3
Ports: 5060 (UDP/TCP), 5061 (TLS)

Media IPs:

54.244.51.0/24
Port Range: 10,000 to 20,000 (UDP)

Europe Ireland Gateways:

Signalling IPs:

54.171.127.192
54.171.127.193
54.171.127.194
54.171.127.195
Ports: 5060 (UDP/TCP), 5061 (TLS)

Media IPs:

54.171.127.192/26
52.215.127.0/24 
Port Range: 10,000 to 20,000 (UDP)

Europe Frankfurt Gateways:

Signalling IPs:

35.156.191.128
35.156.191.129
35.156.191.130
35.156.191.131
Ports: 5060 (UDP/TCP), 5061 (TLS)

Media IPs:

35.156.191.128/25 
3.122.181.0/24 Port Range: 10,000 to 20,000 (UDP)

Asia Pacific Tokyo Gateways:

Signalling IPs:

54.65.63.192
54.65.63.193
54.65.63.194
54.65.63.195
Ports: 5060 (UDP/TCP), 5061 (TLS) 

Media IPs:

54.65.63.192/26
3.112.80.0/24 Port Range: 10,000 to 20,000 (UDP)

Asia Pacific Singapore Gateways:

Signalling IPs:

54.169.127.128
54.169.127.129
54.169.127.130
54.169.127.131
Ports: 5060 (UDP/TCP), 5061 (TLS)

Media IPs:

54.169.127.128/26
3.1.77.0/24 Port Range: 10,000 to 20,000 (UDP)

Asia Pacific Sydney Gateways:

Signalling IPs:

54.252.254.64
54.252.254.65
54.252.254.66
54.252.254.67
Ports: 5060 (UDP/TCP), 5061 (TLS)

Media IPs:

54.252.254.64/26
3.104.90.0/24 Port Range: 10,000 to 20,000 (UDP)

South America Sao Paulo Gateways:

Signalling IPs:

177.71.206.192
177.71.206.193
177.71.206.194
177.71.206.195
Ports: 5060 (UDP/TCP), 5061 (TLS) 

Media IPs:

177.71.206.192/26
18.228.249.0/24 Port Range: 10,000 to 20,000 (UDP)

Glossary

Communications Infrastructure

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

SIP Endpoint

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

SIP URI

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

Twilio SIP Domain

It takes the form {example}.sip.{region}.twilio.com 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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