CCIE Voice Lab 1.8 – SRST and AAR

This lab will challenge the candidate to configure Survivable Remote Site Telephony (SRST) and Automated Alternate Routing (AAR).

CCIE Voice Lab 1.8 Tasks

1. Configure the Los Angeles router for SIP SRST. Phone should display a “SRST Fallback Active” message.

2. While in SRTS mode, calls to 911 or 9-911 should be route immediately. You may only use on outgoing dial-peer to accomplish this task.

3. Preserve the dial-9 for PSTN access during an SRST scenario.

4. While in SRST node, users should be able to dial 4-digit to NY and 7+3… to London. Calls should be routed over the PSTN without the need for users to add prefixes to these locations. You may not use a Prefix in the dial-peer, and this task must be accomplished using a minimal amount of commands.

5. Ensure that users at New York and Los Angeles are able to call each other when bandwidth limits are met.


CCIE Voice Lab 1.8 Solutions

1. Tasks 1 – 4 involved setting up SRST for SIP phones. A very good reference document is the Cisco Unified SIP SRST System Administrator Guide . First, start by configuring the basis SRST commands on the Los Angeles VGWY.

losangeles#
!
voice register global
system message SRST Fallback Active
max-dn 10
max-pool 10
dialplan-pattern 1 2135432... extension-length 4
!
voice register pool 1
translation-profile incoming emergency
id mac 0021.D8B9.BC72
number 1 2001
codec g722-64
!
voice register pool 2
translation-profile incoming emergency
id mac 0021.D8BA.2373
number 1 2002
codec g722-64
!
!
application
global
service alternate default
!
!
ccm-manager fallback-mgcp
!
!
call-manager-fallback
max-conferences 4 gain -6
transfer-system full-consult
!

2. To configure 911 and 9-911 routing, configure 9-911 translation rule and translation profile. Then, apply the translation profile to the voice register pool configurations. Lastly, create a 911 pots dial-peer.

!
voice translation-rule 1
rule 1 /9911/ /911/
!
!
voice translation-profile emergency
translate called 1
!
!
voice register pool 1
translation-profile incoming emergency
!
voice register pool 2
translation-profile incoming emergency
!
!
dial-peer voice 911 pots
description SRST 911
destination-pattern 911
port 0/1/0:23
forward-digits all
!

3. Task 3 and 4 involve the creation of additional pots dial-peers. However, you need to create some additional translation rules and a translation profile to accommodate the 4-digit and 7+4-digit dialing requirements.

!
voice translation-rule 2
rule 1 /\(1...\)/ /212432\1/
rule 2 /^7\(3...\)/ /207654\1/
!
!
voice translation-profile SRST
translate called 2
!
!
dial-peer voice 2 pots
description SRST Long Distance
destination-pattern 91[2-9]..[2-9]......
incoming called-number .
direct-inward-dial
port 0/1/0:23
!
dial-peer voice 3 pots
description SRST Local 10-Digit
destination-pattern 9[2-9]..[2-9]......
port 0/1/0:23
!
dial-peer voice 4 pots
description SRST London
destination-pattern 901144T
port 0/1/0:23
!
dial-peer voice 5 pots
description SRST 4-digit to NY
translation-profile outgoing SRST
destination-pattern 1...
port 0/1/0:23
!
dial-peer voice 6 pots
description SRST 7+4-digit to London
translation-profile outgoing SRST
destination-pattern 73...
port 0/1/0:23
!

4. Test the SIP SRST fallback by shutting down the WAN link from Los Angeles to New York. First, the phones should failover to SRST and display the “SRST Fallback Active” message. The “show voice register all” will provide the SRST configuration. Lastly, place some test calls, using 4-digit, 7+4-digit, and PSTN dial patterns.

losangeles# sh voice register all
VOICE REGISTER GLOBAL
=====================
CONFIG [Version=7.1]
========================
Version 7.1
Mode is srst
Max-pool is 10
Max-dn is 10
Outbound-proxy is enabled and will use global configured value
System message is SRST Fallback Active
timeout interdigit 10
network-locale[0] US (This is the default network locale for this box)
network-locale[1] US
network-locale[2] US
network-locale[3] US
network-locale[4] US
user-locale[0] US (This is the default user locale for this box)
user-locale[1] US
user-locale[2] US
user-locale[3] US
user-locale[4] US

VOICE REGISTER DN
=================

VOICE REGISTER POOL
===================
Pool Tag 1
Config:
Mac address is 0021.D8B9.BC72
Number list 1 : Pattern is 2001
Proxy Ip address is 0.0.0.0
DTMF Relay is disabled
kpml signal is enabled
Translation-profile incoming emergency

Dialpeers created:

dial-peer voice 40003 voip
destination-pattern 2001
redirect ip2ip
session target ipv4:10.1.22.16:5060
session protocol sipv2
digit collect kpml
codec g722-64 bytes 160
after-hours-exempt FALSE

dial-peer voice 40004 voip
destination-pattern 2135432001
redirect ip2ip
session target ipv4:10.1.22.16:5060
session protocol sipv2
digit collect kpml
codec g722-64 bytes 160
after-hours-exempt FALSE

Statistics:
Active registrations : 1

Total SIP phones registered: 1
Total Registration Statistics
Registration requests : 1
Registration success : 1
Registration failed : 0
unRegister requests : 0
unRegister success : 0
unRegister failed : 0


Pool Tag 2
Config:
Mac address is 0021.D8BA.2373
Number list 1 : Pattern is 2002
Proxy Ip address is 0.0.0.0
DTMF Relay is disabled
kpml signal is enabled
Translation-profile incoming emergency

Dialpeers created:

dial-peer voice 40001 voip
destination-pattern 2002
redirect ip2ip
session target ipv4:10.1.22.17:5060
session protocol sipv2
digit collect kpml
codec g722-64 bytes 160
after-hours-exempt FALSE

dial-peer voice 40002 voip
destination-pattern 2135432002
redirect ip2ip
session target ipv4:10.1.22.17:5060
session protocol sipv2
digit collect kpml
codec g722-64 bytes 160
after-hours-exempt FALSE

Statistics:
Active registrations : 1

Total SIP phones registered: 1
Total Registration Statistics
Registration requests : 1
Registration success : 1
Registration failed : 0
unRegister requests : 0
unRegister success : 0
unRegister failed : 0


5. The final tasks require the configuration of setup and configuration of Automated Alternate Routing (AAR). AAR is disabled by default on CUCM, so the first step is to configure the AAR Service Parameters in CUCM by choosing System > Service Parameters > Cisco Call Manager.

Next, configure an AAR group. Since my Adtran PSTN simulator does not accept the “1” for long distance calls, I simply prefix a “9”. In the CCIE Voice Lab, and possibly your home lab, you may need to prefix “91”.

AAR Partitions and Calling Search Spaced are then created for AAR. Below is the NY AAR CSS, which included the NY AAR LD Partition.

Two new Route Lists are created, one for each location. These new route lists used for the AAR only point to the local PSTN gateways. Below is the NY AAR RL.

Create two new route patterns for ten-digit NY to LA and LA to NY calling (remember, my Adtran PSTN simulator does not accept “1+10-digit” dialing). Link these new route patterns to the appropriate AAR Route lists.

Finally, add the appropriate AAR Calling Search Space and AAR Group to each phone and AAR settings for line.

For testing, I temporary modify the bandwidth between locations to 24kbps (one call). Nail up two calls between each location. With the second call, the phone should display “Network Congestion, Rerouting” and the call should be placed via the PSTN.

CCIE Voice Lab 1.7 – Voice CODECs and CAC

Now that the dial plan has been properly configured between New York, Los Angeles, and London, this lab will focus on configuring the proper CODECs between locations as well as Call Admission Control.

CCIE Voice Lab 1.7 Tasks
1. Calls within a location should use G.722 wideband.

2. Calls between locations should use G.729.

3. Allow five concurrent calls between New York and Los Angeles.

4. When calls from London to either New York or LA use the IP WAN, they should negotiate G.729. However, in the event that there are any transcoding issues, only one G.722 call should be allowed across the WAN.


CCIE Voice Lab 1.7 Solutions
The tasks in this lab are relatively simple. A good discussion on locations, regions, and CAC are covered Call Admission Control chapter of Cisco Unified Communications SRND Based on Cisco Unified Communications Manager 7.x. You can also click here and here for some old, but decent Gatekeeper configuration examples.

1. The first step is to create regions for New York, Los Angeles, and London and assign the codecs to be used between each location. You may have done some of these task earlier when configure phones.

2. Next, create Device Pools and assign the Regions accordingly.

3. Next, assign your newly create device pools to devices.

4. Create your locations and assign the appropriate bandwidth. Make sure you understand how CUCM allocates bandwidth for a particular codec. Then, assign these locations to the Phones.

5. On the gatekeeper, configure the proper bandwidth settings.

newyork#
!
gatekeeper
zone local newyork ballplayersllc.com 1.1.1.1
alias static 10.1.10.21 1720 gkid newyork gateway voip ras 10.1.10.21 33072 e164 2002 e164 2001 e164 1003 e164 1002 e164 1001
bandwidth total zone newyork 128
no shutdown

6. Finally, nail up some calls between locations to exceed the bandwidth limits. On CUCM phones, once bandwidth limits are hit, the phones should display “Not Enough Bandwidth”. On the gatekeeper, “debug ras” and “show gatekeeper zone status” are useful commands.

CCIE Voice Lab 1.6 – Call Routing

This lab focuses on developing the Call Routing and dial plan for Ballplayers, LLC. Please note, depending on your PSTN simulation, some task may require digit manipulation not explicitly defined in the tasks. For example, my Adtran does not accept the London international dial plan. Therefore, I will need to strip the 011+44 from the international dialing from New York and Los Angeles. Your solution may vary, depending on what you are using to simulate the PSTN, and what digits that device will accept.

CCIE Voice Lab 1.6 Tasks

1. Ari Gold (1001) and Arliss Michaels (2001) are allowed full dialing privileges, including international. David Wright (1002) and Kobe Bryant (2002) are not allowed international dialing. Finally, Eli Manning can only call on-net calls.

2. New York should be configured with the following call routing options and parameters:

Calls to London: 7+4 digits
Emergency: 911 and 9911
Local: 9+10 digits
Long Distance: 9+1+10 digits
International: 9+011+any number of digits

In addition:

2a. Local calls from New York should user the local VGWY, followed by the Los Angeles VGWY.

2b. Long Distance calls from New York should user the local VGWY, followed by the LA VGWY.

2c. Calls from NY to the LA area code (213) should route via the LA VGWY, followed by the NY VGWY.

2d. International calls to London from NY should use the 7+4 digit pattern via the gatekeeper, followed by the local NY VGWY. Users should not receive secondary dial-tone when calling London extensions.

2e. All other International calls should route via the NY VGWY, followed by the LA VGWY. Eli should be able to call London an on-net call.

3. Los Angeles should be configured with the following call routing options and parameters:

Calls to London: 7+4 digits
Emergency: 911 and 9911
Local: 9+10 digits
Long Distance: 9+1+10 digits
International: 9+011+any number of digits

In addition:

3a. Local calls from LA should user the local VGWY, followed by the NY VGWY.

3b. Long Distance calls from LA should user the local VGWY, followed by the NY VGWY.

3c. Calls from LA to the NY area code (212) should route via the NY VGWY, followed by the LA VGWY.

3d. International calls to London from LA should use the 7+4 digit pattern via the gatekeeper, followed by the local LA VGWY. Users should not receive secondary dial-tone when calling London extensions.

3e. All other International calls should route via the LA VGWY, followed by the NY VGWY. Eli should be able to call London an on-net call.

4. The London dial plan should be configured as follows:

Calls to New York and Los Angeles: 7+4 digits
Emergency: 999 and 9999
Local London: 9+8 digits
Long Distance UK: 9+0+8 digits
International to US: 9+001+any number of digits


5. You may not use any call blocking mechanisms or local routing to accomplish the task in this lab.

6. You may not use the 9.@ NANP wildcard.

7. All external callers should receive the proper E.164 Call ID from the Calling Party.


CCIE Voice Lab 1.6 Solutions
There are three ways to design a dial plan:

• The Traditional Partition/Calling Search Space approach
• The Line/Device CSS approach
• The New Local Route Groups approach

The wording of this lab, in particular task 5 disallowing the use of blocking mechanism and local routing is an attempt to force the user to configure the dial plan using the Traditional Partition/Calling Search Space approach. The dial plan configuration for both New York and Los Angeles is very similar, so the steps below apply to both.

1. First, begin by creating the appropriate partitions for each location.

2. Next, create the Calling Search Spaces (CSS) and assign the Partitions accordingly



3. Route groups are configured next. For each respective route group, only assign the local gateway. For example, for “RG_LA_MGCP_VGWY” only assign the LA VGWY; for RG_NY_H323_VGWY only assign the NY VGWY; and for RG_GKPR only assign gatekeeper trunk configured in the previous lab.

One note about this lab…. During my testing and staging, I ended up removing the Technology Prefix that was required in Lab 1.5. Therefore, you will need to remove this configuration from the trunk as well as the Gatekeeper.

4. The Route Groups are then added to Route Lists. Pay attention to the ordering of the Route Lists in the Route Groups, so that the task requirements are met. For example, below is the ordering for the LA to NY Tail End Hop Off (TEHO) described in the tasks.

5. The Route Patterns are created next, and assigned to the appropriate Route Lists. Here, your configuration may vary slightly from mine, based on your particular PSTN simulator.



For example, in my Adtran only accepts 10-digit dialing (or, at least that is all I could figure it out to accept). Therefore, I apply a “XXXXXXXXXX” mask so that only the trailing ten digits of a call to London is sent. In this example, if a user in NY dials 9-011-44-20-7654-3001, only “20-7654-3001”, which my Adtran then routes to the London PRI. Don’t forget thing such as “Urgent Priority” for 911 and selecting Discard Digits options where appropriate.


6. Lastly, assign the phones and lines to their appropriate partitions in order to honor the Class of Service described in the task.

7. Networking between CUCM and CUCME can be tricky, especially given the tasks as described. For the 7+4-digit dialing, you need to create a translation rule. A rule is also required to add the appropriate prefix to route calls out the PSTN. Finally, for my testing purpose, I create a rule to translate “999” to “911” so that I could test 911 emergency call to my Adtran.

Another helpful hint during testing is to change the preferences of the dial-peer and observe the behavior of the phone class using the “sh voice call summary” command.

!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
h323
!
!
!
voice class codec 1
codec preference 1 g729r8
codec preference 2 g722-64
codec preference 3 g711ulaw
codec preference 4 g711alaw
!
!
translation-rule 1
Rule 1 71 1
Rule 2 72 2
!
!
translation-rule 2
Rule 0 ^999 911
!
!
translation-rule 3
Rule 1 71 2124321
Rule 2 72 2135432
!
!
dial-peer voice 1 pots
description inbound PRI dial-peer
incoming called-number .
direct-inward-dial
port 0/0/0:23
!
dial-peer voice 999 pots
destination-pattern 999
translate-outgoing called 2
port 0/0/0:23
forward-digits 3
!
dial-peer voice 2 pots
description International
destination-pattern 9001T
port 0/0/0:23
!
dial-peer voice 3 pots
description London Local
destination-pattern 9[2-9].......
port 0/0/0:23
!
dial-peer voice 4 pots
description UK Long Distance
destination-pattern 90[2-9].......
port 0/0/0:23
!
dial-peer voice 1000 voip
preference 2
destination-pattern 71...
translate-outgoing called 1
voice-class codec 1
session target ras
dtmf-relay h245-alphanumeric
!
dial-peer voice 2000 voip
preference 2
destination-pattern 72...
translate-outgoing called 1
voice-class codec 1
session target ras
dtmf-relay h245-alphanumeric
!
dial-peer voice 1001 pots
preference 1
destination-pattern 71...
translate-outgoing called 3
port 0/0/0:23
!
dial-peer voice 2001 pots
preference 1
destination-pattern 72...
translate-outgoing called 3
port 0/0/0:23
!