Adding SIP trunk to ISP on CUBE

[carl.townshend]

Hi Guys

When adding a SIP trunk to our ISP on our CUBE, what are the main commands needed, I know we need a dial peer pointing the patterns towards it etc, i see commands like 

registrar dns:voice.ip.com expires 180
sip-server dns:voice.ip.com

what is the registrar used for ?

[M02@rt37]

Hello @carl.townshend 

registrar dns:voice.ip.com expires 180 : This command is used to configure the SIP registrar server to which your CUBE will send registration requests.

The dns:voice.ip.com part indicates the DNS hostname or IP address of the SIP registrar server. The expires 180 parameter sets the registration expiration time to 180 seconds. This means that your CUBE will re-register with the SIP registrar every 180 seconds to keep its registration valid.

sip-server dns:voice.ip.com : This command is another way to specify the SIP registrar server. It sets the SIP server to which the CUBE will send SIP requests. Again, dns:voice.ip.com represents the DNS hostname or IP address of the SIP registrar server.

The purpose of SIP registration is to inform the SIP registrar server about the current location (IP address) of your CUBE. It's commonly used in scenarios where your CUBE has a dynamic IP address, and it needs to periodically update the registrar with its current address.

[carl.townshend]

Hi, 

thanks for the reponse, so do you need a registrar server or can you just have a sip-server?

I am sure I have seen routers configured without the registrar server and just sip server, in what scenario do you have one or the other, or both?

cheers

[Roger Kallberg]

If you need to use a registrar or not depends on your service provider.

[carl.townshend]

Hi Roger, why would they need one or not?

[Shalid Kurunnan Chalil · Accepted solution]

Hi there, 

The following Cisco live presentation will give you a better idea about the SIP Trunk configuration with the service provider. 

Deploying SIP Trunks with Cisco Unified Border Element 

I have taken a few key bits from that and posting here.

1. Information required from Provider:

2. Enabling the CUBE application on your router/gateway:

3. Call routing configuration 

But if you are using registration-based, then the following may help you further. (If your provider requires authentication and registration again built additional protection on unauthorized usage where every outgoing call from CUBE, the ITSP challenges with digest authentication.)

For global level configuration:

SIP-ua

 authentication username <siptrunkitsp> password <password> realm itsp.com

 credentials username <siptrunkitsp> password <password> realm itsp.com

 registrar dns:itsp.com

ref: https://www.youtube.com/watch?v=-R-KHWqAHKQ 

If using a tenant, you can add the following configuration:

voice class tenant 2

  registrar dns: dns.itsp.com expires 3600

  credentials username <siptrunkitsp> password <password> realm itsp.com

  authentication username <siptrunkitsp> password <password> realm itsp.com

  registrar dns:itsp.com

Then, call the tenant on your dial peer:

dial-peer voice 30 voip

voice-class sip tenant 100

session target dns:dns.itsp.com

Ref:

https://www.youtube.com/watch?v=smV-He22MnY

https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/voi-cube-multi-tenants.html

Ref: https://www.cisco.com/c/en/us/solutions/enterprise/interoperability-portal/networking_solutions_products_genericcontent0900aecd805bd13d.html

These references provide detailed configurations. To assist you further, it would be helpful to know the specific details provided by your ITSP, such as whether it involves IP-based authentication or registration-based authentication. you may still need to user or update your sip profile rule depending on your NAT configuration. 

Regards, 

Shalid 

Disclaimer:

Responses are based on personal knowledge and experience. Consider them as guidance. Other members may offer different perspectives or better approaches. No responsibility is assumed for outcomes; discretion is advised.

[carl.townshend]

This is very helpful, thanks for this

[Roger Kallberg]

It depends on if the SP uses registered SIP trunk.

[Carlo Poggiarelli]

Hi Carl

As @Roger Kallberg mentioned, some SP dedicate a IP connection for Voice service so they don't need to autenticate your calls. What they will ask for sure is that the calling number would match the  assigned public DDI range and this could be achieved both on CUCM RP or applyng SIP Profiles on your outgoing DP.

HTH

Regards

Carlo