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SIP Trunking questions [Continued]

prasanna.k1
Level 1
Level 1

The following questions are the continuation of the questions I already asked here.
 

SIP Trunking can be deployed:

  1. For routing the SIP calls on the CUCM cluster to a particular gateway (when the CUCM may be in London and the gateway may be in Chennai)
  2. For connecting to the service provider 

In the deployment #1 (See attachment 1), the call manager cluster and the connected IP phones can be in one of the branch office and the VoIP gateways can be in various regions and connects to appropriate region's PSTN (either via PRI or SIP). While configuring SIP trunks in CUCM, in the destination IP in the SIP Information section the respective gateway's IP address needs to be given (though the gateway have CUBE update or not). We can configure multiple gateways in the same section and the dial plan will/should be having the information on what calls to be routed to which gateway. In this kind of deployment, SIP trunking is a logical connection between the CUCM cluster and the VoIP gateway when the devices are in different geo locations. Service provider cannot restrict the number of parallel SIP sessions between CUCM cluster and VoIP gateway but the bandwidth does. 

In the deployment #2 (See attachment 2), the VoIP gateway will be connected to the service provider via SIP trunk (again a logical connection only via the internet bandwidth provided by the service provider). In case of a SIP call to an analog phone, the VoIP gateway will be routing the call to the service provider via the SIP trunk and the provider will be routing the call to PSTN (unlike PRI directly does the same from VoIP gateway). Here the company have to buy SIP sessions from the service provider in addition to the internet service purchased. There may be some restrictions with the number of parallel calls made. 

 

Correct me if the above understanding is wrong. 

Also please help by clarifying the following questions:

  1. For the deployment scenario #1, what would be the ways we can retrieve the configured SIP trunks through CLI. I knew some of the approaches for retrieving either with SNMP/AXL. It would be of great help if any CLI commands suggested for the same
  2. For the deployment scenario #2, what would be the commands for getting the following:
  • List of trunks configured?  
  • Calls at given point of time (Hope "show call active voice brief" will do)
  • The bandwidth utilized for SIP calls (Hope the approx bandwidth can be calculated with the codec in the "show call active voice brief"  command response)

Please help by clarifying the understanding/assumptions. It would be of great help. 

 

Thanks,
Prasanna

 

1 Accepted Solution

Accepted Solutions

Vivek Batra
VIP Alumni
VIP Alumni

You are absolutely right for both of the deployment scenarios.

1. In deployment#1, you only configure SIP trunk in CUCM whereas in gateway, no SIP trunk. In gateway,  you only configure the dial-peers (with session protocol sipv2) which tells gateway to route the call to CUCM using SIP.

2. show sip-ua commands will give you desired output.

As you already mentioned, depends on codec being invoked for a call, bandwidth will be utilized. For instance, G.711 will utilize 64 Kbps (w/o other layers overhead).

 

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5 Replies 5

Vivek Batra
VIP Alumni
VIP Alumni

You are absolutely right for both of the deployment scenarios.

1. In deployment#1, you only configure SIP trunk in CUCM whereas in gateway, no SIP trunk. In gateway,  you only configure the dial-peers (with session protocol sipv2) which tells gateway to route the call to CUCM using SIP.

2. show sip-ua commands will give you desired output.

As you already mentioned, depends on codec being invoked for a call, bandwidth will be utilized. For instance, G.711 will utilize 64 Kbps (w/o other layers overhead).

 

Thank you Vivek for your response.

So, with the dial-peer configuration, the gateway will be using the SIP trunk to route the in-bound calls to the CUCM and CUCM will use the gateway IP and dial plan to route the calls to the appropriate gateway.

I will explore "sip-ua"  commands on gateway side for scenario#2. It would be of great help  if you can clarify the commands I can use on CUCM side to get the list of SIP trunks configured on CUCM.

Thanks once again!
Prasanna

 

So, with the dial-peer configuration, the gateway will be using the SIP trunk to route the in-bound calls to the CUCM and CUCM will use the gateway IP and dial plan to route the calls to the appropriate gateway.

In IOS, this kind of peering (with session protocol sipv2 command configured under dial-peer) is generally not referred as SIP trunk. However what you have mentioned is right regarding the call flow.

When CUCM receives this call, it will first match the gateway (or SIP trunk) and then check the inbound routing configuration. Generally, you configure inbound CSS on SIP trunk to route the call to desired destination.

 

 It would be of great help  if you can clarify the commands I can use on CUCM side to get the list of SIP trunks configured on CUCM.

CUCM has well defined and intuitive web GUI for configuration and status. You don't need to use CLI interface/commands for regular administrative task.

 

Thank you Vivek :)

Regards,
Prasanna

You are most welcome.

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