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Moving to SIP/do I or do I not need an SBC

Mark Verwey
Level 1
Level 1

My company is talking about moving to SIP trunks and dropping our ISDN/PRI circuits.  We have been in conversation with several carriers about SIP trunks, and are getting mixed answers about the need for an SBC on our side.  One says we do, another says we don't need one.  THey both are terminating on our side to an Ethernet connection which will be plugged into my gateway router.  I understand that I will need to have voip dial-peers configured in order for the router to understand what they are and where to route them.  The thing I'm struggling with is the gateway (Cisco 2921 router) configuration, and want to make sure I understand what I also need to add to my CUCM in order for this to work flawlessly.  Obviously all of the Cisco documents I can find about SIP trunks are stating I need an SBC, in particular they suggest the CUBE license which isn't a big deal I just need to understand why one says we need it and the other says we don't.

Thanks for your advise and assistance in advance,

Mark

1 Accepted Solution

Accepted Solutions

Building direct trunk from CUCM to provider might not be a good idea as provider will have access to the inside network subnet where your CUCM is residing.

Just enable below and add bind commnds to make it work as SIP and set dial-peer to provider .

voice service voip

allow connections sip to sip

SIP

bind control source-interface GigabitEthernetX/X
 bind media source-interface GigabitEthernetX/X.

You can also have bind command specific to the dial-peers so that WAN interface is used for provider side and LAN interface is used for cucm side .

View solution in original post

12 Replies 12

It entirely depends upon the requirements / features you need for SBC. Below are the setup methods you can use
1- SIP Trunk directly pointing from CUCM to the SIP Provider for the calls
2- SIP Trunk from the CUCM to the VG and then an ISDN circuit t the provider 
3- CUCM >> Cisco SBC [CUBE] >> SIP Trunk >> Provider. 
So, Ideally you should be able to use the 1st method.depending upon the provider if they support and allow [They should be as I have seen it as per my experience]

HTH 
Regards
Abhay

Regards
Abhay Singh Reyal
The Only Way To Do Great Work Is To Love What You Do. If You Haven’t Found It Yet, Keep Looking. Don’t Settle

Sounds like your first method might work, however how do I configure the gateway to do this.

Having a direct SIP trunk between CUCM and the SIP provider might be technically feasible, but the CUCM team has stated that they do not support doing that, they require a CUBE or SBC in between.

HTH

java

if this helps, please rate

Deepak Mehta
VIP Alumni
VIP Alumni

You may not need a CUBE to get it working and SIP gateway should just work fine .However if you have CUBE licenses enabled it will enable more features.

You can set it up  redundant pair with stateless fail over with call preservation ,also  latest IOS version has many features like more flexibility when you are doing SIP header modifications on CUBE.

Destination server group - You can have multiple destination under single outbound dial-peer

Multiple incoming patterns under same dial-peer  or you can also use a text file this way you don't need multiple dial-peers.

You can use 2921 as cube( paper based license no need to install any file etc )  however based on it CPU and memory it will support max 200-400 active sessions .So you may want to check on the utilization before you plan on using 2921.

Sounds like the Cube license is more of an honor system with Cisco, being there is no license or software to install on the actual device.  I don't have any issues purchasing the license if needed that's why I'm asking for advise. 

I am running IOS ver 15.3(3)M7, not sure its CPU, but it has 512 mb of memory and PVDM DIMM with 256 channels. 

What or how do I configure the router/gateway to do the following:

1.  Act as a SIP gateway. 

2.  If I need to add the CUBE license, can I turn it on (and how) get it working before I actually receive the paper from my reseller?

Is there a Cisco document that goes over in great detail on how to configure the 2921 gateway router to use SIP.

Building direct trunk from CUCM to provider might not be a good idea as provider will have access to the inside network subnet where your CUCM is residing.

Just enable below and add bind commnds to make it work as SIP and set dial-peer to provider .

voice service voip

allow connections sip to sip

SIP

bind control source-interface GigabitEthernetX/X
 bind media source-interface GigabitEthernetX/X.

You can also have bind command specific to the dial-peers so that WAN interface is used for provider side and LAN interface is used for cucm side .

THanks for this suggestion, I really think this is the way Im going to go for now.  We are starting a trial/demo next week, I have a totally seperate gateway that I will be using so as not to interfear with the existing voice traffic.

You stated under the SIP setting add the 2 bind statements

which one points to the provider and which one points to my CUCM?

I typically do it per dial-peer assuming I have an external/DMZ interface and a seperate LAN interface

 

dial-peer voice 3 voip
 translation-profile outgoing strip7
 destination-pattern 7T
 session protocol sipv2
 session target ipv4:X.X.X.X
 session transport udp
 voice-class codec 1  
 voice-class sip early-offer forced
 voice-class sip profiles 100
 voice-class sip tenant 10
 voice-class sip bind control source-interface GigabitEthernet0/0/1
 voice-class sip bind media source-interface GigabitEthernet0/0/1
 dtmf-relay rtp-nte
 no vad
dial-peer voice 4 voip
 preference 1
 destination-pattern .T
 session protocol sipv2
 session target ipv4:X.X.X.X
 session transport udp
 voice-class codec 1  
 voice-class sip bind control source-interface GigabitEthernet0/0/0
 voice-class sip bind media source-interface GigabitEthernet0/0/0
 dtmf-relay rtp-nte
 no vad

Thanks for your reply,

This SIP trunk is going to be installed on a 2951 router, that is on a private network.  The Gi0/0 port is on the same network as my CUCM, the gi0/1 port is on a /30 network range connected directly with the SIP provider. 

 

Gotcha. Then just bind to the gi0/1 interface on dial-peers toward the provider and gi0/0 on dial-peers towards CUCM.

Perfect.  I think this will definately work now without having to add an SBC into the mix, my provider already has one in place.

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