Voice Deep Dive Workbook

Version 3.5

Module 12

Cisco Unified Border Element: CUBE :: Tasks

Lab Instructions:
Prior to starting this lab, ensure that you have both read through the Voice Rack Rental Guide
before beginning this lab, so that you are fully aware of how all equipment works and is
accessed, and that you have loaded both the initial router/switch configs from INEs members
site, and that you have manually loaded the CUCM initial configs as described below.

Note
- Most Deep Dive lab scenarios require existing configuration before they can begin to be
configured. Also, most lab scenarios use different pre-configuration from other Deep Dive
labs. Therefore it is necessary to load the "Startup Configs" prior to beginning any of these lab
scenarios.
- To load your router and switch configs, first log into your INE.com Members account, then
navigate to the "Rack Rental" tab, and click on "Control Panel" >> "Click here to choose a
configuration to be loaded on your Voice Rack", then choose the appropriately named
selection for this module. This will only load your router and switch configs (pstn, r1, r2, r3,
sw1, sw2).
- To load your CUCM server initial configs, please log into the CUCM Admin WebUI, and use
the Bulk Administration tool to upload & import the appropriately named DD-ModXXCUCM-Startup.tar" file that was included in your Members account for this module. Here are
two brief demonstration videos on how to successfully accomplish this upload and import into
CUCM: Importing or Exporting Configurations in CSV file from CUCM (15 mins), and Fixing
Any Issues That May Arise With Importing CSS and CdPTP & CngPTPs (8 mins).

Copyright 2011 Internetwork Expert

www.INE.com
-1-

2011 Internetwork Expert, Inc. All rights reserved.

INE, the INE logo, and Voice Deep Dive are registered trademarks of Internetwork Expert, Inc. in the United States and certain other countries.
All other trademarks mentioned in this document or Website are the property of their respective owners. The use of the word partner does not imply a
partnership relationship between Internetwork Expert and any other company.

Voice Deep Dive Workbook

Version 3.5

Module 12

12.1 SIP Trunk to AT&T ITSP for All CorpHQ PSTN Call Routing

Provision your CUCM Cluster to provide both inbound and outbound PSTN calling
services to all Local, National and International numbers (no emergency services) for
phones at the CorpHQ site via a new AT&T ITSP SIP Trunk, as their primary routing
option. AT&T will now be routing all inbound calls to CorpHQ phone DID's via this SIP
trunk primarily as well.
Provision a demarcation point between your CUCM Cluster and the AT&T SIP Trunk
using the CorpHQ router
AT&T has assigned you FQDNs to use for connectivity that provide for redundancy
and load balancing of your calls to them, and they are listed below. Also, a few more
FQDNs have been pre-provisioned for you that define things such as the CorpHQ
router. Ensure you use these FQDNs, that AT&T sees calls coming from your CorpHQ
FQDN (not IP address in SIP messages), and that all calls are load balanced to AT&T.
The DNS server and FQDNs are as follows:
o DNS Server IP Address:
177.1.10.110
o AT&T IP FlexReach FQDNs:
sip1.att.com
o
sip2.att.com
o INE CorpHQ-R1 FQDN:
corphqr1.ine.com
Ensure that IP's from both the "Inside" and the "Outside" of the CUBE are hidden
(CUCM, IP Phones, etc are hidden from AT&T, and vice-versa that AT&T's IP
addresses are hidden from the CUCM and IP Phones)
AT&T has told you to send them all calls over this SIP Trunk in the Full +E.164 format
(include +)
Ensure that a backup route to the PSTN via ISDN on the CorpHQ router is still fully
functional

Copyright 2011 Internetwork Expert

www.INE.com
-2-

2011 Internetwork Expert, Inc. All rights reserved.

INE, the INE logo, and Voice Deep Dive are registered trademarks of Internetwork Expert, Inc. in the United States and certain other countries.
All other trademarks mentioned in this document or Website are the property of their respective owners. The use of the word partner does not imply a
partnership relationship between Internetwork Expert and any other company.

Voice Deep Dive Workbook

Version 3.5

Module 12

12.2 Conforming to ITSP Requirements :: Various SIP-Attributes

Ensure that any error messages from CUCM are passed through, intact, to AT&T
Ensure CLID and CNAM are sent to AT&T and seen on the PSTN Phone
Ensure SIP signaling integrity is maintained end-to-end across CUBE throughout the
course of the call
Ensure G.729 Annex B is allowed to be negotiated by CUBE and is interoperable with
CUCM, otherwise AT&T tells you that calls will fail
AT&T has informed us that they will only support RFC2833 as a DTMF Relay type, and
typically use the payload value of 100, although it is possible that a value of 96 could
be sent. Ensure that DTMF Relay is negotiated properly, and that even if AT&T
happens to send an RTP value of 96, that DTMF works for PSTN users calling into the
CUCM and any VM or IVR they might encounter.
AT&T has informed us that they will only respond to a call setup if the SDL is sent in
the SIP INVITE header, therefore ensure that this is supported, however ensure CUCM
is not forced to allocate a MTP to support this feature request

12.3 Conforming to ITSP Requirements :: SIP Header Conversions

After this trunk has been functioning for a few days, you being to notice that if a call
goes on for longer than 30 minutes, that it drops exactly at the 30 minute mark. After
you troubleshoot this by setting up a live call, you find that exactly at that 30 minute
mark, AT&T is sending you a SIP UPDATE message, to which the CUCM doesn't
respond well to, since all mid-call messaging is passed through. To solve this issue,
configure CUBE to instruct AT&T that you do not support SIP UPDATE messages
Your company has an engineering department that sometimes has a need to perform
tests that involve calls that can sometimes last many hours at a time. Determine what
value CUCM is passing to (and ultimately through) CUBE as the maximum call
duration, and then configure CUBE to inform AT&T that each call should be allowed to
stay active for a maximum of 24 hours. Do not modify any parameters in CUCM to
accomplish this task.

12.4 Advance Call Admission Control Mechanisms with CUBE

Configure CUBE to send the "User-Busy" flag with a reason indicating a lack of
resources, back to any party sending CUBE a SIP INVITE message, when any of the
following conditions are met:
o If the router's CPU rises above 90%, block all calls until it falls below 50%
o If the router's I/O memory rises above 80%, block all calls until it falls below 65%
o If there is a sudden spike of SIP INVITE messages - where the router receives
more than 50 INVITES and/or more than 8 calls in less than a 100ms sliding
window of time - block all calls until it falls below that value

Copyright 2011 Internetwork Expert

www.INE.com
-3-

2011 Internetwork Expert, Inc. All rights reserved.

INE, the INE logo, and Voice Deep Dive are registered trademarks of Internetwork Expert, Inc. in the United States and certain other countries.
All other trademarks mentioned in this document or Website are the property of their respective owners. The use of the word partner does not imply a
partnership relationship between Internetwork Expert and any other company.

Voice Deep Dive Workbook

Version 3.5

Module 12

12.5 Skype SIP Trunk for Branch2 Site

Being quite happy with your existing SIP Trunks, you now wish to incorporate the
usage of the low-cost, high performance option of Skype to your Branch2 site in
Amsterdam. Provision another CUBE element at the Branch2 site and switch outbound
call routing for that site accordingly
Ensure that any error messages from CUCM are passed through, intact, to AT&T
Ensure CLID and CNAM are sent to AT&T and seen on the PSTN Phone
Ensure SIP signaling integrity is maintained end-to-end across CUBE throughout the
course of the call
Ensure that calls from phones at the Branch2 site now use the CUBE to Skype trunk
as their primary route for all outbound Local, National and International calls. Skype
cannot assume ownership of the Branch2 site's existing ISDN DID range of phone
numbers, however it has already been provisioned with a new number local to
Amsterdam and is already configured to send inbound calls to the new range of
numbers listed below:
o +312075745XX
Skype and the Branch2 router already have FQDNs provisioned and they are as
follows:
o INE Branch1-R2 FQDN:
branch2r3.ine.com
o Skype Business SIP Trunk FQDN:
sip.skype.com

Copyright 2011 Internetwork Expert

www.INE.com
-4-

2011 Internetwork Expert, Inc. All rights reserved.

INE, the INE logo, and Voice Deep Dive are registered trademarks of Internetwork Expert, Inc. in the United States and certain other countries.
All other trademarks mentioned in this document or Website are the property of their respective owners. The use of the word partner does not imply a
partnership relationship between Internetwork Expert and any other company.