SIP and SIP Trunking overview

Use SIP when you need to connect PBX infrastructure, enterprise SBCs, cloud telephony providers, or existing SIP trunks. SIP Trunking is the common production pattern for routing calls between your telephony core and Voximplant-managed Voice AI scenarios.

Voximplant supports SIP calling in three distinct models:

SIP trunking based on number-based routing
User-based SIP routing where you create a Voximplant user for each SIP endpoint and route based on username
SIP registration where Voximplant registers on external SIP infrastructure and is treated like an extension

Handling SIP calls with external SIP Infrastructure

Security and allow-listing

If your SIP infrastructure uses a fixed public IP, add it to the SIP allow-list in Voximplant security settings. For firewall configuration, use the official Voximplant SIP/media IP sources below:

SIP trunking with number-based routing

SIP trunking gateway model

Use this when your PBX/trunk forwards calls by dialed number.

Create routing rule for destination pattern

In your Voximplant application, create a routing rule whose pattern matches the inbound destination number format you expect - i.e. ^\\+1[2-9]\\d{9}$.

Attach Voice AI scenario to the rule

Attach your inbound Voice AI scenario to that rule. See Configure Voximplant - Create a scenario for details.

Forward SIP calls from PBX/trunk to Voximplant

Route calls to: number@application.account.voximplant.com where number matches the routing-rule pattern, application is your Voximplant application name, and account is your Voximplant account name. Alternatively, you can also use this format app_id#account_id#destination@sip.voximplant.com where destination must be a number and app_id and account_id are the numeric ids found in your application settings.

Allow-list source IPs if needed

If your PBX has fixed public IPs, allow-list them in Settings -> Security -> SIP White List in manage.voximplant.com. SIP White List in Security settings

User-based SIP routing (Voximplant user per endpoint)

Use this when each SIP endpoint should authenticate as a distinct user and route by username or a pattern based on usernames.

Create one Voximplant user per SIP endpoint (agent seat, extension, or softphone identity).

Open Users in your application

In manage.voximplant.com, go to Applications, open your app, then open Users. Create a Voximplant user

Create user credentials for each SIP endpoint

Create one user per endpoint (for example: support01, support02) and set a strong password. Keep this username/password pair because SIP clients will use it for registration. User editor with username and password

Create routing rule by username

Create routing rules that match the target username or alias and attach the relevant scenario. See Configure Voximplant - Create a scenario for details.

Login from your SIP software with these credentials

Login to this account on your SIP software using user@application.account.voximplant.com and the Voximplant user password. SIP software authorization example You can specify UDP/TCP transport, or specify TLS transport to secure your connection. Some SIP infrastructure may require that you specify the password and options in the address: sip:username:password@domain:5061;transport=tls

Place calls to that user

Now you can place calls with your SIP software to and from that user.

If you automate provisioning, you can also create users via the Management API: AddUser.

SIP registration (Voximplant as registered extension)

Use this when Voximplant should register to external SIP infrastructure (such as a PBX) and behave like a phone connected to that infrastructure.

Create extension/user in external SIP system

In your PBX or SIP provider, create the extension / ID Voximplant will use.

Create SIP registration in Voximplant

In Voximplant settings, create a SIP registration with that extension’s domain, username, and password. Add SIP registration in manage.voximplant.com

Attach SIP registration to application

Attach the SIP registration to your application. This is available from the SIP Registrations tab in the left pane. When you do this, you will need to select the desired routing rule that handles calls to this registration. Note that you do not need to configure the rule pattern in this case — the rule works for all the calls to this SIP registration.

Check the registration status

Open SIP Registrations in Control Panel and verify the registration shows as connected/active before testing calls. You should see s status of Success if the registration is active. You can also verify status via Management API: GetSipRegistrations.

Place calls to this registration

Now you can place calls to and from this registration just like you would with any other phone / device connected to your SIP infrastructure.

Pricing note

Please note that you pay for SIP registrations monthly according to our price list. So if you create a SIP registration and delete it immediately, your account is charged for the SIP registration monthly fee.

When a SIP registration status changes from Successful to Failed and vice versa, you get an email notification to the contact email addresses and the HTTP callback invokes. You can learn more about HTTP callbacks in this article.

Placing outbound SIP calls from Voximplant

Use VoxEngine.callSIP() for outbound SIP dialing.

Check registration status

Before placing outbound calls through a SIP registration, confirm the registration is active in Control Panel (SIP Registrations) or via GetSipRegistrations.

How to make calls with SIP registration

To place an outbound SIP call through a registration, pass regId in the third argument of VoxEngine.callSIP(...). You can find regId in the SIP registration list in Control Panel, or in GetSipRegistrations and CreateSipRegistration responses. Usually you only need to specify the destination username/number of the called party.

Outbound SIP via URI

1 const call = VoxEngine.callSIP("sip:1001@pbx.example.com", "+15557654321");

Outbound SIP via registration (regId)

1 const regId = 1234567; // from SIP Registrations list or HTTP API response
2 const destination = "1001"; // usually extension/username in your SIP system
3 const call = VoxEngine.callSIP(`sip:${destination}`, "+15557654321", { regId });

For programmable outbound launching, start the routing rule from GUI or call startScenarios and pass custom data (for example destination/caller ID) to the scenario.

Example script_custom_data

1 {"callerid":"+15557654321","number":"1001"}

SIP Voice AI connector example:

Ultravox SIP trunking guide