NOTE: Programmable SIP is currently in Developer Preview. In order to get set up with Programmable SIP, please contact Support or your Account Manager.
Vonage’s Programmable SIP enables you to integrate your existing SIP Infrastructure with Vonage’s powerful conversational communications platform. This integration will enable you to connect to mobile, landline, SIP and WebRTC endpoints with minimal work, including browsers and mobile applications. It will also bring Voice API functionality, such as multichannel recording, IVR, Text to Speech, WebSocket connectivity for AI integrations, and the power of contextual conversations to your platform.
A Vonage SIP Domain contains configuration you need to connect to Vonage SIP endpoints and link to your Vonage application.
To route a SIP call to your Vonage application, you need to create a unique Vonage domain, such as
yourcompany. The domain name will form the SIP URI, for example
sip:firstname.lastname@example.org, and any calls routed to that SIP URI will be routed to your application NCCO
The authentication method will be determined within your configuration of the Vonage domain. Vonage will authenticate the request and forward it to your application.
Some example domains:
yourcompany.sip-eu.nexmo.com email@example.com firstname.lastname@example.org
You can configure an Access Control List for your domain so that your Vonage application will only accept calls from specific endpoints and devices. You do this by adding their IP addresses to an allow list.
Calls made to a Programmable SIP domain must be handled at a regional level. You must use a Request URI with a regional domain. Please be aware that a Request URI without a regional component in the domain will fail the call.
The following code will indicate to Vonage that you want this SIP call to be handled in the EU:
Available domains are the same as the A records:
sip-us.nexmo.com: USA sip-eu.nexmo.com: Europe sip-ap.nexmo.com: Asia Pacific
You can specify any additional headers you need when sending a SIP Request. Any headers provided must start with
X- and will be sent to your
answer_url with a prefix of
SipHeader_. For example, if you add a header of
X-UserId with a value of
1938ND9, Vonage will add
SipHeader_X-UserId=1938ND9 to the request made to your
CAUTION: Headers that start with
X-Nexmo are not sent to your
You can use the Voice API NCCO connect action to connect a call to your SIP endpoints. The detailed documentation is here.
The SIP Connect feature support will continue where you can dial your virtual number via your SIP endpoint that is attached to your application. Digest Authentication is the accepted authentication method for SIP Connect.
To test this functionality have your PBX forward calls to
sip.nexmo.com. Here is an example of doing so with an Asterisk extension, transmitting a custom header that will be sent to your
exten => 69100,1,SIPAddHeader(X-UserId:ABC123) exten => 69100,2,Dial(SIP/nexmo/14155550100)