Making SIP Calls
Learn to initiate a VoIP session and connect to your SIP-enabled endpoints with Twilio Programmable Voice by using the Twilio REST API to send a POST request to the Calls resource. You can use this guide to create an outbound call center, deploy AI-powered agents, and support PSTN connectivity.
See Related reference documentation to learn more about the SIP endpoint parameters used in this guide.
Initiate SIP sessions via the REST API by POSTing to the same calls resource used to make outbound phone calls. Once the call is connected, Twilio will then fetch the TwiML you specify for the call. For example, make a SIP call by POSTing to your account's calls list resource URI:
/2010-04-01/Accounts/{AccountSid}/Calls
All outgoing call features and parameters are supported — the only difference is that you pass different values in the "To" and "From" parameters. In the "To" parameter, put the SIP URI you are trying to connect to. In the "From" parameter, specify the user you want to show up in the From header in the SIP request.
You must POST the following parameters:
| Parameter | Description |
|---|---|
| To | The SIP URI to which you want to connect |
The 'To' parameter specifies a SIP address for Twilio to connect to. The body of the URI element should be a valid SIP URI under 255 characters. For example:
sip:michael@example.com
Pass headers in the To parameter by appending them to the end of the SIP URI. For certain sdks, & will need encoding as %26. The total characters passed in a header must be under 1024. For example:
sip:michael@example.com?mycustomheader=foo&myotherheader=bar
Set a parameter on your SIP URI to specify what transport protocol you want to use. You may use UDP, TCP or TLS. By default, Twilio sends your SIP INVITE over UDP. Change this by using the transport parameter:
sip:jack@example.com;transport=tcp
Also, when using transport=tls, this will only encrypt SIP signaling messages and not RTP. To use SRTP and to encrypt SIP signaling, please add a parameter secure=true to your SIP URI:
sip:jack@example.com;secure=true
You may POST the following parameters:
| Parameter | Description |
|---|---|
| From | This value is used to populate the username portion of the From header that is passed to the SIP endpoint. This may be any alphanumeric character, as well as the plus, minus, underscore, and period characters (+-_.). No spaces or other characters are allowed. |
| SipAuthUsername | Your authentication username. |
| SipAuthPassword | The password for the user. |
Basic SIP dial using the REST API.
1// Download the helper library from https://www.twilio.com/docs/node/install2const twilio = require("twilio"); // Or, for ESM: import twilio from "twilio";34// Find your Account SID and Auth Token at twilio.com/console5// and set the environment variables. See http://twil.io/secure6const accountSid = process.env.TWILIO_ACCOUNT_SID;7const authToken = process.env.TWILIO_AUTH_TOKEN;8const client = twilio(accountSid, authToken);910async function createCall() {11const call = await client.calls.create({12from: "Jack",13to: "sip:kate@example.com",14url: "http://www.example.com/sipdial.xml",15});1617console.log(call.sid);18}1920createCall();
Response
1{2"account_sid": "ACXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX",3"answered_by": null,4"api_version": "2010-04-01",5"caller_name": null,6"date_created": "Tue, 31 Aug 2010 20:36:28 +0000",7"date_updated": "Tue, 31 Aug 2010 20:36:44 +0000",8"direction": "inbound",9"duration": "15",10"end_time": "Tue, 31 Aug 2010 20:36:44 +0000",11"forwarded_from": "+141586753093",12"from": "Jack",13"from_formatted": "(415) 867-5308",14"group_sid": null,15"parent_call_sid": null,16"phone_number_sid": "PNaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",17"price": "-0.03000",18"price_unit": "USD",19"sid": "CAaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",20"start_time": "Tue, 31 Aug 2010 20:36:29 +0000",21"status": "completed",22"subresource_uris": {23"notifications": "/2010-04-01/Accounts/ACaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Calls/CAaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Notifications.json",24"recordings": "/2010-04-01/Accounts/ACaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Calls/CAaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Recordings.json",25"payments": "/2010-04-01/Accounts/ACaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Calls/CAaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Payments.json",26"events": "/2010-04-01/Accounts/ACaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Calls/CAaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Events.json",27"siprec": "/2010-04-01/Accounts/ACaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Calls/CAaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Siprec.json",28"streams": "/2010-04-01/Accounts/ACaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Calls/CAaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Streams.json",29"transcriptions": "/2010-04-01/Accounts/ACaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Calls/CAaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Transcriptions.json",30"user_defined_message_subscriptions": "/2010-04-01/Accounts/ACaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Calls/CAaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/UserDefinedMessageSubscriptions.json",31"user_defined_messages": "/2010-04-01/Accounts/ACaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Calls/CAaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/UserDefinedMessages.json"32},33"to": "sip:kate@example.com",34"to_formatted": "(415) 867-5309",35"trunk_sid": null,36"uri": "/2010-04-01/Accounts/ACaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Calls/CAaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa.json",37"queue_time": "1000"38}
Pass user and password to your SIP call for authentication.
1// Download the helper library from https://www.twilio.com/docs/node/install2const twilio = require("twilio"); // Or, for ESM: import twilio from "twilio";34// Find your Account SID and Auth Token at twilio.com/console5// and set the environment variables. See http://twil.io/secure6const accountSid = process.env.TWILIO_ACCOUNT_SID;7const authToken = process.env.TWILIO_AUTH_TOKEN;8const client = twilio(accountSid, authToken);910async function createCall() {11const call = await client.calls.create({12from: "Jack",13sipAuthPassword: "secret",14sipAuthUsername: "jack",15to: "sip:kate@example.com",16url: "http://www.example.com/sipdial.xml",17});1819console.log(call.sid);20}2122createCall();
Response
1{2"account_sid": "ACXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX",3"answered_by": null,4"api_version": "2010-04-01",5"caller_name": null,6"date_created": "Tue, 31 Aug 2010 20:36:28 +0000",7"date_updated": "Tue, 31 Aug 2010 20:36:44 +0000",8"direction": "inbound",9"duration": "15",10"end_time": "Tue, 31 Aug 2010 20:36:44 +0000",11"forwarded_from": "+141586753093",12"from": "Jack",13"from_formatted": "(415) 867-5308",14"group_sid": null,15"parent_call_sid": null,16"phone_number_sid": "PNaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",17"price": "-0.03000",18"price_unit": "USD",19"sid": "CAaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",20"start_time": "Tue, 31 Aug 2010 20:36:29 +0000",21"status": "completed",22"subresource_uris": {23"notifications": "/2010-04-01/Accounts/ACaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Calls/CAaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Notifications.json",24"recordings": "/2010-04-01/Accounts/ACaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Calls/CAaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Recordings.json",25"payments": "/2010-04-01/Accounts/ACaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Calls/CAaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Payments.json",26"events": "/2010-04-01/Accounts/ACaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Calls/CAaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Events.json",27"siprec": "/2010-04-01/Accounts/ACaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Calls/CAaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Siprec.json",28"streams": "/2010-04-01/Accounts/ACaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Calls/CAaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Streams.json",29"transcriptions": "/2010-04-01/Accounts/ACaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Calls/CAaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Transcriptions.json",30"user_defined_message_subscriptions": "/2010-04-01/Accounts/ACaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Calls/CAaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/UserDefinedMessageSubscriptions.json",31"user_defined_messages": "/2010-04-01/Accounts/ACaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Calls/CAaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/UserDefinedMessages.json"32},33"to": "sip:kate@example.com",34"to_formatted": "(415) 867-5309",35"trunk_sid": null,36"uri": "/2010-04-01/Accounts/ACaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Calls/CAaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa.json",37"queue_time": "1000"38}
Pass headers to your SIP Dial as part of the SIP URI.
1// Download the helper library from https://www.twilio.com/docs/node/install2const twilio = require("twilio"); // Or, for ESM: import twilio from "twilio";34// Find your Account SID and Auth Token at twilio.com/console5// and set the environment variables. See http://twil.io/secure6const accountSid = process.env.TWILIO_ACCOUNT_SID;7const authToken = process.env.TWILIO_AUTH_TOKEN;8const client = twilio(accountSid, authToken);910async function createCall() {11const call = await client.calls.create({12from: "Jack",13to: "sip:kate@example.com?X-hatchkey=4815162342",14url: "http://www.example.com/sipdial.xml",15});1617console.log(call.sid);18}1920createCall();
Response
1{2"account_sid": "ACXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX",3"answered_by": null,4"api_version": "2010-04-01",5"caller_name": null,6"date_created": "Tue, 31 Aug 2010 20:36:28 +0000",7"date_updated": "Tue, 31 Aug 2010 20:36:44 +0000",8"direction": "inbound",9"duration": "15",10"end_time": "Tue, 31 Aug 2010 20:36:44 +0000",11"forwarded_from": "+141586753093",12"from": "Jack",13"from_formatted": "(415) 867-5308",14"group_sid": null,15"parent_call_sid": null,16"phone_number_sid": "PNaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",17"price": "-0.03000",18"price_unit": "USD",19"sid": "CAaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",20"start_time": "Tue, 31 Aug 2010 20:36:29 +0000",21"status": "completed",22"subresource_uris": {23"notifications": "/2010-04-01/Accounts/ACaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Calls/CAaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Notifications.json",24"recordings": "/2010-04-01/Accounts/ACaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Calls/CAaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Recordings.json",25"payments": "/2010-04-01/Accounts/ACaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Calls/CAaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Payments.json",26"events": "/2010-04-01/Accounts/ACaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Calls/CAaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Events.json",27"siprec": "/2010-04-01/Accounts/ACaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Calls/CAaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Siprec.json",28"streams": "/2010-04-01/Accounts/ACaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Calls/CAaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Streams.json",29"transcriptions": "/2010-04-01/Accounts/ACaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Calls/CAaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Transcriptions.json",30"user_defined_message_subscriptions": "/2010-04-01/Accounts/ACaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Calls/CAaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/UserDefinedMessageSubscriptions.json",31"user_defined_messages": "/2010-04-01/Accounts/ACaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Calls/CAaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/UserDefinedMessages.json"32},33"to": "sip:kate@example.com?X-hatchkey=4815162342",34"to_formatted": "(415) 867-5309",35"trunk_sid": null,36"uri": "/2010-04-01/Accounts/ACaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa/Calls/CAaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa.json",37"queue_time": "1000"38}
This guide teaches the basics required for the following use cases:
You can use this guide to provision a programmatic outbound infrastructure that dials destination networks and bridges conversations directly over your existing SIP-based infrastructure. This configuration cuts down on public telecommunication routing expenses and maintains centralized session control.
To learn more advanced features that you can use with outbound call centers, see Voice outbound call centers.
You can use this guide to connect conversational AI agents directly to your internal communications infrastructure via SIP. By streaming real-time audio from your SIP trunks or PBX systems to voice AI endpoints, you can deploy intelligent digital agents capable of managing complex customer service operations natively within your existing telephony ecosystem.
To learn more advanced features that you can use with AI-powered agents, see Voice AI agents.
You can use this guide to bridge the gap between legacy infrastructure and the public telephone network. By utilizing Twilio to route calls from your private SIP endpoints or IP-PBX systems out to standard mobile networks and landlines globally, you gain highly scalable, reliable, and cost-effective PSTN connectivity.
To learn more advanced features that you can use with PSTN connectivity, see Voice PSTN connectivity.
After following this guide, you can establish an active SIP VoIP connection using the Twilio REST API. You have successfully configured custom authentication parameters, adjusted SIP header payloads inside your URI syntax, and established standard fallback endpoints to gracefully handle session timeouts or structural routing failures.
Explore the following guides to build on what you've learned in this guide:
- Gather user input by using the keypad (DTMF Tones): Use the
<Gather>verb to build interactive telephone menus. - Accept payments during calls using Stripe: Integrate Stripe with Twilio to securely process credit card payments over the phone.
- Modify calls in progress: Dynamically update ongoing telephone sessions.