Seperating SIP traffic based on site.

[Kleid Gjataj]

We have  CUCMs 9.1 in 3 locations

CUCM_Loc1

CUCM_Loc2

CUCM_Loc3

We have a SIP trunk between CUCM and third party IVR. The third party IVR needs to see the SIP invites come from each individual  CUCM to handle the audio locally and not traverse the WAN.

The GWs are MGCP and are setup to use the local CUCM first and then rollover to the other sites.

GW_Loc1 --> CUCM_Loc1---->IVR_Loc1

The issue is that all the SIP invites to the third party IVR come from the first server in the CUCM Group and not from the CUCM server at the location. Therefor the IVR software doesn't know to use the server at the location, causing RTP to traverse the WAN.

Any ideas would be appreciated.

[Ayodeji Okanlawon]

The first thing to note is that RTP shouldn't be terminated on your CUCM. The question then is why is the IVR streaming media to your CUCM servers? Do you have "require MTP" checked on the sip trunk?

What version of CUCM are you using?

Once we can eliminate RTP going to your CUCM, your media traffic will be directed to the endpoints at the local sites.

[Kleid Gjataj]

RTP is not going thru CUCM. RTP flows directly from GW to IVR servers. The IVR servers decide on which server to use based on the "From" portion of the SIP invite. 

Since the Invite is always coming from the first server in the CUCM Group, only one IVR server gets used.

[Ayodeji Okanlawon · Accepted solution]

First of all, the IVR server cant send media to the From header in a SIP INVITE. Media is sent to the IP address in the connection info header in the SDP of either your INVITE or your response. This is what the connection info looks like

c=IN IP4 10.1.6.17

So something is not right in that statement.

Secondly, you can influence which cucm server is used to make outbound calls ie (the from header) based on a few factors..

1.  Enable run on all active nodes in your cucm sip trunk

2. Enable run on all active cucm nodes on the route list..

With these two enabled, this will ensure that the cucm that is used to send the call out the trunk is not going to be the one on the route group of the sip trunk or route list but the one that the device making the call is registered to. So if phone A from location A registered to cucm1 makes a call, the from header will show cucm1.

You can refer to this document which details the process of selecting a cucm node that will be used for outbound calls on a sip trunk

https://supportforums.cisco.com/blog/12088366/sip-trunks-and-run-all-active-cm-nodes