CUCM 12.0 Centralized site model design question

mubashirathar1999 · ‎09-25-2019

Hi fellows,

I am a beginner at Collaboration technologies and will really appreciate if you guys can help me out in setting up a voice infrastructure setup at my work project. So we have 2 office sites, one in US and one in Canada (Total of less than 1000 users). We are migrating our Voip network from Nortel CS1K to CUCM 12.0. I will be using the following design:

1) Cisco BE6K hardware with CUCM 12.0, CUC, IMP and EXP C/E - 1 BE6K will be deployed each site

2) CUCM Pub and Sub 1 in Site#1 and Sub 2 in Site#2 with CUBE as VGW for SIP trunking to PSTN

3) For Site#1 , the CUCM Group consist of CUCM Pub and Sub 1 (Sub 1 will be handling Call Control) and for Site# 2 the CUCM Group consist of CUCM Pub and Sub 2 (so in case of Sub 2 outage, call control will be handled by Pub which is located at a different Geographical site)

4) We would be using Cisco Jabber as a softphone client and for MRA. We want to use Site#1 users to be registered to EXP C/E only on Site#1 BE6K and Site#2 users to be registered on EXP C/E only on Site#2 BE6K. However for Jabber to detect MRA it resolves the collab-edge record to CUCM IP, how can I assign within my single domain network 2 SRV resolutions for the same collab-edge record ?

5) Is there anything else I should be worried about using this design model?

6) Any sample CUBE configs I can use on ISR4331 for NANP dial plans and connection establishing to ISP?

Thank you all for looking into this.

Regards

Anthony Holloway · ‎09-25-2019

Wow, you tagged Jaime Valencia! I can't wait to see his response! Bookmarked!

mubashirathar1999 · ‎09-25-2019

Lol sorry didnt know if that was offensive by any means.Apologies, Edited the post. This is only my second post in the community and the first one never received a response so thought if I may be doing some mistake in posting.

Cheers

Anthony Holloway · ‎09-26-2019

It's not offensive to me. Tagging people is a part of the forum's features. Tag away!

mubashirathar1999 · ‎09-25-2019

@Anthony Holloway

Can you please provide some suggestion on my post above? Thanks and appreciate your response.

Anthony Holloway · ‎09-26-2019

Let's see...

4) We would be using Cisco Jabber as a softphone client and for MRA. We want to use Site#1 users to be registered to EXP C/E only on Site#1 BE6K and Site#2 users to be registered on EXP C/E only on Site#2 BE6K. However for Jabber to detect MRA it resolves the collab-edge record to CUCM IP, how can I assign within my single domain network 2 SRV resolutions for the same collab-edge record ?

Technically the _collab-edge resolves to Expressway-E, so I think you would need to use GeoDNS (https://en.wikipedia.org/wiki/GeoDNS), such that clients in Canada resolve to one site, while the clients in the US resolve to the other. However, this would not limit nor prevent a users from crossing the border and using the opposite site's MRA. The only way I know how that would work is with different domains.

5) Is there anything else I should be worried about using this design model?

There is A LOT to be worried about. 30ms RTT, probably somewhere in the 20Mbps BW range, QoS, region and location design, media resources use and placement, device mobility, E911, or just 911 at all for MRA clients. A LOT!

6) Any sample CUBE configs I can use on ISR4331 for NANP dial plans and connection establishing to ISP?

version 16.6
voice service voip
 mode border-element
 allow-connections sip to sip
 sip
  midcall-signaling passthru media-change
!
voice class uri 1100 sip
 host ipv4:4.4.4.4 ; your provider IP
!
voice class uri 2100 sip
 host ipv4:10.1.1.1 ; your CUCM IP
 host ipv4:10.2.2.2 ; your CUCM IP
!
voice class server-group 2200
 ipv4 10.1.1.1 preference 1 ; your CUCM IP
 ipv4 10.2.2.2 preference 2 ; your CUCM IP
!
voice class sip-options-keepalive 2200
 description CUCM Nodes (Intentionally Left Blank)
!
dial-peer voice 1100 voip
 description Incoming Call Leg From Provider
 session protocol sipv2
 incoming uri via 1100
 voice-class sip bind control source-interface GigabitEthernet0/0/1
 voice-class sip bind media source-interface GigabitEthernet0/0/1
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
!
dial-peer voice 1200 voip
 description Outgoing Call Leg To Provider
 destination-pattern +T
 session protocol sipv2
 session target ipv4:4.4.4.4
 voice-class sip bind control source-interface GigabitEthernet0/0/1
 voice-class sip bind media source-interface GigabitEthernet0/0/1
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
!
dial-peer voice 2100 voip
 description Incoming Call Leg From CallManager
 session protocol sipv2
 incoming uri via 2100
 voice-class sip bind control source-interface GigabitEthernet0/0/0
 voice-class sip bind media source-interface GigabitEthernet0/0/0
 dtmf-relay sip-kpml rtp-nte
 codec g711ulaw
 no vad
!
dial-peer voice 2200 voip
 description Outgoing Call Leg To CallManager
 destination-pattern [2-9]..[2-9]......$
 session protocol sipv2
 session server-group 2200 
 voice-class sip options-keepalive profile 2200
 voice-class sip bind control source-interface GigabitEthernet0/0/0
 voice-class sip bind media source-interface GigabitEthernet0/0/0
 dtmf-relay sip-kpml rtp-nte
 codec g711ulaw
 no vad
!

mubashirathar1999 · ‎09-26-2019

Thank you Anthony, really appreciate that. For outbound dial peer match, do you suggest to use Dial-peer provision policy or keeping it simple by matching destination pattern only?

Anthony Holloway · ‎09-26-2019

I prefer destination pattern matching, because it's not uncommon for me to need to treat a called number differently, and if I used DPGs, then I created a one-off dial-peer with a destination-pattern of my choosing, well, then, it wont work....because DPGs are higher priority match than destination patterns.

https://www.cisco.com/c/en/us/support/docs/voice/ip-telephony-voice-over-ip-voip/211306-In-Depth-Explanation-of-Cisco-IOS-and-IO.html#toc-hId-1969397414

It does make for a clean config...just don't forget to put a destination-pattern on the outgoing dial-peer anyway, otherwise your SIP OPTIONS wont work (none will be sent out).