NOTE: Programmable SIP is currently in Developer Preview. In order to get set up with Programmable SIP, please contact Support or your Account Manager.
Nexmo’s Programmable SIP enables you to integrate your existing SIP Infrastructure with Nexmo’s powerful conversational communications platform. This integration will enable you to easily connect to mobile, landline, SIP and WebRTC endpoints, 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 Nexmo SIP Domain contains configuration you need to connect to Nexmo SIP endpoints and link to your Nexmo application.
To route a SIP call to your Nexmo application, you need to create a unique Nexmo domain, such as
yourcompany. The domain name will form the SIP URI, for example
sip:email@example.com, 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 Nexmo domain. Nexmo will authenticate the request and forward it to your application.
Some example domains:
yourcompany.sip-eu.nexmo.com firstname.lastname@example.org email@example.com
IP addresses of your devices and endpoints can be configured to reach your Nexmo application. Nexmo will only accept calls from the IP address list that is configured for your domain. This method will enable you to whitelist your endpoints’ and devices’ IP Addresses per your application.
If you require your call to be handled by a specific Nexmo region, then you can specify that by using R-URI with such information in the domain part.
The following code will indicate to Nexmo 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, Nexmo 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)