03-28-2013 08:41 AM - edited 03-16-2019 04:30 PM
I am just about to deplot a cucm 8.6 connected to a CUBE gateway running on a 2901.
due to a number of strange operational reasons the SIP provider also has to provide their own gateway connected to the SIP trunk. so logically we will have two CUBE gateways conneted back to back - one controlled by the CUCM and the other connected to the SIP trunk.
I presume the above is supported and will function normally?
any special config required? I preusme the session targets dont change just the default route to get there
Thanks
03-28-2013 09:00 AM
Sure, not problems here, just ensure your routing between them is handled properly i.e. static route, routing protocol, direct connection, etc.
HTH,
Chris
03-28-2013 09:22 AM
Agree with Chris (+5) There shouldne be any problems with this setup...I put together a best practice guide for a new sip deplyment..Have a look and hopefully it wil help you with your deployment
Here are some thoughts for you
1. Configure CUBE for media flow through. In this Mode CUBE acts as a true B2BUA. Advantages you get include address hiding and security becaue CUBE terminates and re-originate both signalling and Media. In this mode CUBE becomes a point of demarcation from th external world.
2. Configure CUCM to use Delayed Offer and CUBE to convert delayed offer to ealry offer...This prevents the need for you to use MTP to send Early offer on CUCM
voice service voip
early-offer forced
3. Configure DTMF signalling method on sip trunk to "No preference" This setting allows Unified CM to make an optimal decision for DTMF and to minimize MTP allocation.
4. Configure your CUBE to meet the requirements of your ITSP. Ask if they have configuration templates or any specific configuration they like you to use. This will save you time troubleshooting. Most of them dont use the default port 5060 because of security, confirm with your proivider what ports they use.
voice service voip
allow-connections sip to sip
sip
early-offer forced
header-passing
error-passthru
5. Use SIP to SIP...Use end to end sip. CUCM---sip---CUBE--sip----ITSP
6. Create a Trusted list of IP addresses on your CUBE is your CUBE IOS is 15.1 .2(T) and above.
voice service voip
ip address trusted list
ipv4 x.x.x.x.255.255.255.255
ipv4 y.y.y.y 255.255.255.0
7. Configure your inbound and outbound dial-peer approriately
Inbound Dial-Peer for calls from CUCM to CUBE (CUCM sending 9 +all digits dialled to CUBE)
dial-peer voice 100 voip
description *** Inbound calls from CUCM***
incoming called-number 9T
session protocol sipv2
codec g711ulaw
dtmf-relay rtp-nte
Outbound Dial-Peer for calls from CUBE to CUCM (SP will be sending 10 digits inbound)
dial-peer voice 200 voip
description *** Calls to Primary CUCM***
destination-pattern [2-9].........
session protocol sipv2
session target ipv4:
codec g711ulaw
dtmf-relay rtp-nte
Note: If more than 1 CUCM cluster exists, you will have to create multiple such ial-peers with “preference” for CUCM redundancy
Inbound Dial-Peer for calls from SP to CUBE
dial-peer voice 100 voip
description *** Inbound Calls from ITSP ***------------------(catch-all for all inbound PSTN calls)
incoming called-number [2-9].........
session protocol sipv2
codec g711ulaw
dtmf-relay rtp-nte
Outbound Dial-Peer for calls from CUBE to SP
dial-peer voice 200 voip
description *** Outbound PSTN Calls***
destination-pattern 9[2-9].........
session protocol sipv2
voice-class sip bind control source gig0/1
voice-class sip bind media source gig0/1
session target ipv4:
codec g711ulaw
dtmf-relay rtp-nte
NB: The bind command is only appripriate if you are using a single interface to connect to CUCM and ITSP. If you have two differnt interfaces then dont use any bind commands. Using a bind command in this scenario can lead to one-way audio etc
8. SIP Normalization:
You may need to configure sip normnalization to modify sip headers, CLI etc. A good example is during call forwarding. You may need to change your diversion headers to match the CLI your provider is expecting.. if your call forwarding is failing during testing this may be the reason..We can help you with this.
9. Media Resources
Plan your solution properly. Consider if you will need Xcoders, MTP, Conference bridge etc. You may avoid the need for xcoders if you confure your regions properly and use voice class codecs on your sip profiles. It is important to know if there are any endpoints in your network that do not support dtmf relay rtp-nte. You can avoid the use of MTP if you configure your dial-peers to have multiple dtmf types for thos phones that do not support rtp-nte
e.g
dial-peer voice 1 voip
session protocol sipv2
dtmf-relay rtp-nte digit-drop sip-kpml (if your phones support kpml..then this will be used)
If in your environment you will need to do xcoding or CFB then ensure you have PVDMS
10.FAX
If you have FAX in your network, determine what fax protocol your sip provider supports. Dont assume. Ask them and confirm in writing what they support. I have seen legal cases because of fax failures over sip trunks
Configure your FAX devices in seperate device pools and use porefix to route calls using G711 only. Even if you are using T38, ensure your fax use G711 to establish the voice calls
Finally
11. Have a detailed and carefully planned TEST Plan. Test the FF:
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"opportunity is a haughty goddess who waste no time with those who are unprepared"
03-28-2013 12:10 PM
thanks for such a comprehensive reponse. some things I never thought about.
in my setup
CUCM---CUBE1----CUBE2---ITSP
can I just a check of my dial peers are correct?
ITSP IP: 50.50.50.50
CUCM: 80.80.80.80
CUBE 1: (ip 10.10.10.10)
dial-peer voice 100 voip
description *** INBOUND TO CUCM ***
session protocol sipv2
destination-pattern [1-9].........
session target ipv4:80.80.80.80
incoming called-number .
voice-class codec 2
dtmf-relay rtp-nte
no vad
dial-peer voice 200 voip
description *** Outbound PSTN Calls***
destination-pattern 9T
session protocol sipv2
voice-class sip bind control source gig0/1
voice-class sip bind media source gig0/1
session target ipv4:50.50.50.50
codec g711ulaw
dtmf-relay rtp-nte
ip route 50.50.50.50 255.255.255.255 20.20.20.20
CUBE 2: (ip 20.20.20.20)
Same Dial Peers as above
ip route 80.80.80.80 255.255.255.255 10.10.10.10
03-28-2013 02:38 PM
Ian,
It looks okay. The only question I have is this: Is CUCM able to talk to CUBE on 10.10.10.10? Reason is ecause I can see that you are putting your bind ocmmands on this interface? If CUCM cant talk to this ip address, they you will have some issues
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"opportunity is a haughty goddess who waste no time with those who are unprepared"
04-02-2013 07:08 AM
had another question,
Cube 1 will be at customers location but the CUCM is located off site- how do I ensure the media is terminated /
anchored on the CUBE 1?
04-02-2013 07:57 AM
Ian,
This is your call flow:
ITSP---CUBE1--CUBE2---CUCM---IP Phone
You are goind to have RTP terminated on both CUBEs
ITSP----RTP---CUBE1--RTP---CUBE2--------RTP--IP PHone
This is how your media will flow
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"opportunity is a haughty goddess who waste no time with those who are unprepared"
04-02-2013 08:35 AM
so if not using an MTP the media is automatically terminated on the CUBE no special config required
04-02-2013 08:44 AM
Yes Ian. Thats absolutely correct. The other scenario where media is not terminated on CUBE is when you have configured media flow-around in which case only signalling will be terminated on CUBE. The default behaviiour is media flow-through where media is terminated on CUBE
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"opportunity is a haughty goddess who waste no time with those who are unprepared"
04-10-2013 07:36 AM
another quick one - for pstn calls from PSTN to IP phone its like this
ITSP----RTP---CUBE1--RTP---CUBE2--------RTP--IPphone
when the call routes to voicemail the media is then termnated on that server
ITSP----RTP---CUBE1--RTP---CUBE2--------RTP--VM?
with attendant console the media will termintte on the AC server until the call is answerd by an phone
ITSP----RTP---CUBE1--RTP---CUBE2--------RTP--AC,
then (when call is answered)
ITSP----RTP---CUBE1--RTP---CUBE2--------RTP--IPPhone
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