The <Dial>
verb's <Sip>
noun lets you set up VoIP sessions by using SIP — Session Initiation Protocol. With this feature, you can send a call to any SIP endpoint. Set up your TwiML to use the <Sip>
noun within the <Dial>
verb whenever any of your Twilio phone numbers are called. If you are unfamiliar with SIP, or want more information on how Twilio works with your SIP endpoint, please see the SIP overview.
The SIP INVITE message includes the API version, the AccountSid
, and CallSid
for the call. Optionally, you can also provide a set of parameters to manage signaling transport and authentication, or configure Twilio to pass custom SIP headers in the INVITE message: this method includes headers such as UUI (User-to-user Information).
Once the SIP session completes, Twilio requests the <Dial>
action URL, passing along the SIP CallID header, the response code of the invite attempt, any X-headers passed back on the final SIP response, as well as the standard Twilio <Dial>
parameters.
If Enhanced Programmable SIP Features is not enabled on your account, only one <Sip>
noun may be specified per <Dial>
, and the INVITE message may be sent to only one SIP endpoint. Also, you cannot add any other nouns (eg <Number>
, <Client>
) in the same <Dial>
as the SIP. If you want to use another noun, set up a callback on the <Dial>
to use alternate methods.
To specify the geographic region from which Twilio will send SIP-out traffic towards your communication infrastructure, you must include the region
parameter in your SIP URI. For example, if the region=ie1
parameter is included in your SIP URI, Twilio will send the SIP traffic from the Europe Ireland region:
_10<?xml version="1.0" encoding="UTF-8"?>_10<Response>_10 <Dial>_10 <Sip>sip:alice@example.com;region=ie1</Sip>_10 </Dial>_10</Response>
Region | Location |
---|---|
us1 | North America Virginia |
us2 | North America Oregon |
ie1 | Europe Ireland |
de1 | Europe Frankfurt |
sg1 | Asia Pacific Singapore |
jp1 | Asia Pacific Tokyo |
br1 | South America São Paulo |
au1 | Asia Pacific Sydney |
If the region
parameter is not specified, Twilio will send SIP-out traffic from the North America Virginia region.
All of the existing <Dial>
parameters work with the <Sip>
noun (record, timeout, hangupOnStar, etc). For SIP calls, the callerId attribute does not need to be a validated phone number. Enter any alphanumeric string. Optionally include the following chars: +-_.
, but no whitespace.
Within the <Sip>
noun, you must specify a URI for Twilio to connect to. The URI should be a valid SIP URI under 255 characters. For example:
Send username and password attributes for authentication to your SIP infrastructure as attributes on the <Sip>
noun.
Attribute Name | Values |
---|---|
username | Username for SIP authentication |
password | Password for SIP authentication |
For example:
Send custom headers by appending them to the SIP URI — just as you'd pass headers in a URI over HTTP. For example:
While the SIP URI itself must be under 255 chars, the headers must be under 1024 characters. Any headers starting with the x-
prefix can be sent this way.
You can also send multiple parameters and values as part of the x-
header
_10<?xml version="1.0" encoding="UTF-8"?>_10<Response>_10 <Dial>_10 <Sip>sip:jack@example.com?x-customname=Madhu%2CMathiyalagan%3BTitle%3DManager&x-myotherheader=bar</Sip>_10 </Dial>_10</Response>
UUI (User-to-User Information) header can be sent without prepending x-
_10<?xml version="1.0" encoding="UTF-8"?>_10<Response>_10 <Dial>_10 <Sip>sip:jack@example.com?User-to-User=123456789%3Bencoding%3Dhex&x-myotherheader=bar</Sip>_10 </Dial>_10</Response>
The following standard SIP headers can also be sent without prepending x-
Remote-Party-ID
P-Preferred-Identity
P-Called-Party-ID
_10<?xml version="1.0" encoding="UTF-8"?>_10<Response>_10 <Dial>_10 <Sip>sip:bob@example.com?x-foo%3Dbar&User-To-User=foobar&Remote-Party-ID=%3Csip%3Afoo%40example.com%3E%3Bparty%3Dcalling&P-Preferred-Identity=%3Csip%3Afoo%40example.com%3E&P-Called-Party-ID=%3Csip%3Afoo%40example.com%3E</Sip>_10 </Dial>_10</Response>
Set a parameter on your SIP URI to specify what transport protocol you want to use. Currently, this is limited to UDP
, TCP
and TLS
. By default, Twilio sends your SIP INVITE over UDP
. Change this by using the transport parameter:
Alternatively, you may customize it to use TLS for SIP signaling. When using TLS, the default port will be 5061 however, a different port may be specified.
When a SIP call is answered, Twilio passes the following parameters with its request in addition to the standard TwiML [Voice request parameters][request parameters]:
Parameter | Values |
---|---|
Called | To header of the SIP Invite message. The SIP identifier of the called party. |
Caller | From header of the SIP Invite message. The SIP identifier of the party that initiated the call. |
SipCallId | The SIP call ID header of the request made to the remote SIP infrastructure. |
SipDomain | The host part of the SIP request. |
SipDomainSid | Your SIP Domain ID. It is 34 characters long, and always starts with the letters SD . |
SipHeader_ | The name/value of any X-headers returned in the 200 response to the SIP INVITE request. This is applicable only if you are using SIP [custom headers][custom-headers]. |
SipSourceIp | Source IP address for SIP signaling. |
When you invoke [dial action][dial-action] attribute and <Sip>
, Twilio passes the following parameters with its request in addition to the standard [dial action][dial-action] parameters. Use the action callback parameters to modify your application based on the results of the SIP dial attempt:
Parameter | Values |
---|---|
DialSipCallId | The SIP call ID header of the request made to the remote SIP infrastructure. |
DialSipResponseCode | The SIP response code as a result of the INVITE attempt. |
DialSipHeader_ | The name/value of any X-headers returned in the final response to the SIP INVITE request. |
The <Sip>
noun supports the following attributes that modify its behavior:
Attribute Name | Allowed Values | Default Value |
---|---|---|
method | GET , POST | POST |
[password][authentication] | Password for SIP authentication | |
statusCallbackEvent | initiated , ringing , answered , completed | none |
statusCallback | any url | none |
statusCallbackMethod | GET , POST | POST |
url | call screening url | none |
[username][authentication] | Username for SIP authentication | |
machineDetection | Enable , DetectMessageEnd | None |
machineDetectionTimeout | 3 -60 | 30 |
machineDetectionSpeechThreshold | 1000 -6000 | 2400 |
machineDetectionSpeechEndThreshold | 500 -5000 | 1200 |
machineDetectionSilenceTimeout | 2000 -10000 | 5000 |
amdStatusCallback | Any URL | None |
amdStatusCallbackMethod | GET , POST | POST |
The url
attribute allows you to specify a url for a TwiML document that
runs on the called party's end, after they answer, but before the two parties are
connected. You can use this TwiML to privately <Play>
or <Say>
information to the
called party, or provide a chance to decline the phone call using <Gather>
and <Hangup>
. If [answerOnBridge][dial-answer-on-bridge] attribute is used on <Dial
>,
the current caller will continue to hear ringing while the TwiML document executes on the other end.
TwiML documents executed in this manner are not allowed to contain the <Dial>
verb.
The method
attribute allows you to specify which HTTP method Twilio should
use when requesting the URL specified in the url
attribute. The default is POST
.
When dialing out to a number using <Dial>
, an outbound call is initiated. The
call transitions from the initiated
state to the ringing
state when the
phone starts ringing. It transitions to the answered
state when the call is
picked up, and finally to the completed
state when the call is over. With
statusCallbackEvent
, you can subscribe to receive webhooks for the different
call progress events: initiated
, ringing
, answered
, or completed
for a
given call.
The statusCallbackEvent
attribute allows you to specify which events Twilio
should webhook on. To specify multiple events separate them with a space:
initiated ringing answered completed
. If a statusCallback
is provided and no
status callback events are specified the completed
event will be sent by default.
As opposed to creating an outbound call via the API, outbound calls created
using <Dial>
are initiated right away and never queued. The following shows a
timeline of possible call events that can be returned and the different call
statuses that a <Dial>
leg may experience: