Zentrunk is Plivo’s SIP Trunking service that provides global coverage for your outbound and inbound voice calls. More than just a regular SIP Trunk, Zentrunk works with your current cloud or on-premise communications infrastructure. Whether you’re looking to increase capacity of your current telecom stack, increase coverage and phone number inventory, or extend your on-premise infrastructure to the cloud, Zentrunk can get you started instantly. No minimum spends, carrier negotiations or long-term contracts to maintain, Zentrunk lets you provision SIP Trunks instantly through our user interface.

Before you get started, you will need to have the following:

  • A Plivo account - Click here to signup for free.
  • IP-PBX or SBC with an internet connection. For more information, see our interconnection guides.

Outbound SIP Trunking

Outbound SIP Trunking

Zentrunk’s Outbound SIP Trunks enable you to reach fixed and mobile phones in 200+ countries. There are no limitations or restrictions on channels or ports. Every trunk comes with unlimited concurrent call capacity. Get all the standard features of a Telco (i.e., dynamic CLI, DTMF support, per second billing) and more (e.g., secure trunking, fraud detection, instant provisioning). You only pay for what you use, with no long term contracts.

Create an Outbound Trunk

In order to use Zentrunk to terminate calls, you need to setup authentication for the trunk on Plivo Console. Authentication of your Trunk ensures that Zentrunk only accepts traffic sent securely by your infrastructure. You can configure your trunk to be authenticated with an IP Group or an Auth Group, or both. For more information on creating an IP Group and Auth Group, see steps 5 to 9 in the section below.

To create an Outbound Trunk

  1. On the Trunks page, click Outbound Trunk.
  2. On the Outbound Trunks page, click Create New Trunk.
  3. In the New Trunk window, enter a name for your trunk (for example, Plivo Test).
    Note: The trunk is Enabled by default.
  4. Under Authentication, select the IP Group or the Auth Group.
    Note: Make sure you select either an IP Group or an Auth Group. The IP Group provides the list of IP addresses from which the SIP Invite will be accepted for this trunk. The Auth Group provides a username and password that will be used to authenticate the SIP Invite. Create Outbound Trunk

  5. To add a new IP Group, click Add new IP Group.
  6. On the Create IP Group window, enter a name for your IP Group (for example, TestACL), and then enter the IP addresses to be whitelisted in the IP Address List field. Note: You can add multiple IP addresses separated by a comma.
  7. Click Create IP Group to save and add your IP Group Create IP Group

  8. To add a new Auth Group, click Add new Auth Group.
  9. On the Create Auth Group window, enter a name for your Auth Group (for example, TestAuthGroup), the Username, and the Password.
  10. Click Create Auth Group to save and add your Auth Group. Create Auth Group

  11. Once you have selected the IP Group and Auth Group, click Create Trunk. Your Outbound trunk will be created.

Inbound SIP trunking

Inbound SIP Trunking

With Zentrunk, you can have instant access to inbound phone number DIDs in Plivo’s over 5 million phone number inventory representing over 60 countries. Each phone number comes provisioned with unlimited concurrent call capacity. Zentrunk customers can also instantly search, filter, and provision fixed, mobile, toll-free, and SMS enabled phone numbers through the Zentrunk API or user interface. Our carrier team is also standing by to help you port your phone numbers to Zentrunk from your current provider.

Create an Inbound Trunk

To create an Inbound Trunk:

  1. On the Trunks page, click Inbound Trunk tab.
  2. On the Inbound Trunks page, click Create New Inbound Trunk.
  3. On the New Trunk window, enter a name for your trunk (for example, Plivo SIP Trunk).
    Note: The Enabled checkbox is selected by default.
  4. Select the Primary URI and Fallback URI of your PBX.
    Note: The Primary URI is the FQDN or IP address to which all calls are forwarded first. If the Primary URI is unresponsive, the calls will be forwarded to the Fallback URI. Create Inbound Trunk

  5. To add a new Primary or Fallback URI, click Add new URI.
  6. On the Create URI window, enter a name for your URI (for example, TestURI), and then enter the URI (the FQDN or IP Address of your VoIP infrastructure).
  7. Click Create URI to save and add your URI. Create URI

  8. Once you have selected your Primary and Fallback URI, click Create Trunk. Your inbound trunk will be created.

Assigning an inbound trunk to a Phone Number

Once you have created and configured your Inbound trunk, assign the inbound trunk to a Phone number. To assign an Inbound Trunk

  1. On the Product Navigation bar, click PHONE NUMBER.
  2. On the Numbers page, under YOUR NUMBERS, click the phone number you wish to use for the Inbound Trunk.
  3. In the EDIT NUMBERS window, select Zentrunk from the APP TYPE list.
  4. From the Trunk list, select the Trunk you wish to use with the phone number, and then click UPDATE. Assign Zentrunk

To know more about buying a Plivo Phone number, please navigate to the Buy a number section.

IP Address Whitelisting

You may need to whitelist Plivo IPs in your firewall to ensure that calls get routed without interruption. Please ensure that you whitelist all the IPs mentioned as the calls might get routed through a different region in an unlikely event of service disruption in a specific region.

The following are the list of IP addresses that you would need to whitelist.

Regions IP Addresses Signalling Ports Media Ports
North California, USA 5060 (UDP/TCP)
5061 (TLS)
10000 - 30000 (UDP/TCP)
Virginia, USA 5060 (UDP/TCP)
5061 (TLS)
10000 - 30000 (UDP/TCP)
Frankfurt, Germany 5060 (UDP/TCP)
5061 (TLS)
10000 - 30000 (UDP/TCP)
Sao Paulo, Brazil 5060 (UDP/TCP)
5061 (TLS)
10000 - 30000 (UDP/TCP)
Sydney, Australia 5060 (UDP/TCP)
5061 (TLS)
10000 - 30000 (UDP/TCP)
Singapore 5060 (UDP/TCP)
5061 (TLS)
10000 - 30000 (UDP/TCP)

Supported SIP methods

Zentrunk currently supports the following SIP methods: ACK, BYE, CANCEL, INVITE and UPDATE.