Documentos de Académico
Documentos de Profesional
Documentos de Cultura
Volume 1
Course Introduction
1
Overview
1
Learner Skills and Knowledge
3
Course Goal and Objectives
4
Course Flow
5
Additional References
5
Cisco Glossary of Terms
6
Your Training Curriculum
1-1
Multisite Deployment Implementation
L Overview
1-1
1-1
Module Objectives
1-3
Identifying Issues in a Multisite Deployment
1-3
Objectives
1-4
Multisite DeploymentIssues Overview
1-6
Quality Issues
1-7
Quality Issues Example: Jitterand Packet Drops
1-8
Bandwidth Issues
1-8
Example: Bandwidth Issues
Bandwidth Issues Example: Voice and DataTraffic Competing for Bandwidth 1-9
Bandwidth Issues Example: Load Caused by Centralized Media Services 1-10
1-11
Availability Issues
1-12
Availability Issues Example: IP WAN Failure
1-13
Dial Plan Issues
Example: Variable-Length Numbering, E.164 Addressing, and DID 1-16
1-17
Fixed vs. Variable-Length Numbering Plans
Detection of End of Dialing in Variable-Length Numbering Plans 1-18
Optimized Call Routing and PSTN Backup 1-20
1-21
Example: TEHO
Example: Overlapping and Nonconsecutive Numbers 1-22
jm 1-23
Various PSTN Requirements
Issues Caused by Different PSTN Dialing 1-25
ImplementingCisco Unified Communications Manager. Part 2 (CIPT2)v8.0 )2010 Cisco Systems, Inc.
Implementing TEHO ]_J24
Considerations for Using Remote PSTN Gateways - l-i«
TEHO Example Without Local Route Groups 1"127
mm TEHO Example with Local Route Groups 1"129
Implementing Globalized Call Routing ]-] 31
Globalized Call Routing: Number Formats 1-133
Normalization ofLocalized Call Ingress onGateways 1-136
mm Normalization of Localized Call Ingress from Phones 1-137
Localized Call Egress atGateways 1-138
Localized Call Egress at Phones 1-140
&. Globalized Call-Routing Example: Emergency Dialing 1-142
mm Considering Globalized Call-Routing Interdependences 1-146
Globalized Call Routing—TEHO Advantages 1-147
Globalized Call Routing—TEHO Example 1-148
* Summary I'ltn
mm References I Icl*
Module Summary ,
References 1-151
fk. Module Self-Check 1'153
'§m Module Self-Check Answer Key 1'158
Centralized Call-Processina Redundancy Implementation 2^
m- Overview 2-1
Me Module Objectives 2-1
Examining Remote Site Redundancy Options 2^
^. Objectives 2"3
^^ Remote Site Redundancy Overview 2-4
^m Remote Site Redundancy Technologies 2-5
When to Use MGCP Fallback 2-6
When to Use Cisco Unified SRST 2-7
f When to Use Cisco Unified Communications Manager Express inSRST Mode 2-9
•* Cisco Unified SRST Operation 2-10
Cisco Unified SRST Function: Switchover Signaling 2-11
Cisco Unified SRST Function: Call Flow After Switchover 2-12
f Cisco Unified SRST Function: Switchback 2-13
** Cisco Unified SRST Timing 2-14
MGCP Fallback Operation 2-15
MGCP Gateway Fallback: Switchover 2-16
* MGCP Gateway Fallback: Switchback 2-17
mm MGCP Gateway Fallback Process 2-18
Cisco Unified SRST Versions and Feature Support 2-19
Cisco Unified SRST 8.0 Platform Density 2-20
f' Plus (+) Prefix and E.164 Supportin Cisco Unified SRST 2-21
*•' Support for Multiple MOH Sources 2-22
Dial Plan Requirements for MGCP Fallbackand Cisco Unified SRST Scenarios 2-23
Ensure Connectivity for Remote Sites 2-24
** Ensure Connectivity from Main Site UsingCFUR 2-25
*•* CFUR Considerations 2-26
CFUR Interaction with Globalized Call Routing 2-28
CFUR Example WithoutGlobalized Call Routing 2-30
"* CFUR Example with Globalized Call Routing 2-31
<•*» Keeping Calling Privileges Active in SRST Mode 2-32
Cisco Unified SRST Dial Plan Requirements Example 2-33
Summary 2-34
* References 2-34
© 2010 Cisco Systems, Inc. Implementing Cisco Unified Communications Manager, Part2(CIPT2) v8.0
Implementing SRST and MGCP Fallback 2-35
Objectives 2-35
MGCP Fallbackand Cisco Unified SRST Configuration Overview 2-36
MGCP Fallback and Cisco Unified SRST Configuration Requirements 2-37
Cisco Unified Communications ManagerSRST Configuration 2-38
Step 1: SRST Reference 2-39
Step 2: Device Pool 2-40
Cisco IOS Gateway SRST Configuration 2-41
Steps 1 and 2: Enabling Cisco Unified SRST and Setting Cisco Unified SRST IP Address 2^2
Steps 3 and 4: Setting Maximum Directory Numbers and Telephones 2-43
Steps 5 and 6: Setting Maximum Directory Numbers Per Phone and Keepalive Timer 2^J4
Cisco Unified SRST Configuration Example 2-45
Cisco IOS GatewayMGCP GatewayFallback Configuration 2-46
Steps 1 and 2: Enabling MGCP Fallback and Setting Fallback Service 2-47
MGCP Fallback Configuration Example 2-48
Cisco Unified Communications Manager Dial Plan Configuration for SRSTSupport 2-49
Step 1: Define a CSS for CFUR 2-49
Step 2: Setting Max Forward Unregistered Hops to DN 2-50
Step 3: Configuring CFUR 2-51
Cisco IOS GatewayMGCP Fallback and Cisco Unified SRST Dial Plan Configuration 2-53
Additional SRST Dial Plan Requirements 2-54
Cisco Unified SRST Dial Plan Commands: Dial Peer 2-56
Cisco Unified SRST Dial Plan Commands: Open Numbering Plans 2-60
Cisco Unified SRST Dial Plan Commands: Number Modification (Voice Translation Profiles) 2-62
Cisco Unified SRST Dial Plan Commands: Number Modification (Voice Translation Rules) 2-63
Cisco Unified SRST Dial Plan Commands: Number Modification (Profile Activation) 2-64
Cisco Unified SRST Dial Plan Commands: COR 2-65
Cisco Unified SRST Dial Plan Example 2-67
Summary 2-70
References 2-70
Implementing Cisco Unified Communications Manager Express in SRST Mode 2-71
Objectives 2-71
Cisco Unified Communications Manager Express Overview 2-72
Cisco Unified Communications Manager Express in SRST Mode 2-73
When to Use Cisco Unified Communications Manager Express in SRST Mode 2-74
Cisco Unified Communications Manager Express Features 2-78
Important Cisco Unified Communications Manager Express Features 2-79
General Configuration of Cisco Unified CommunicationsManager Express 2-80
Cisco Unified Communications Manager Express: Basic Configuration Example 2-82
Providing Phone Loads 2-83
Cisco Unified Communications Manager Express: MOH 2-84
Additional MOH Sources 2-85
Additional Music on Hold Sources—Configuration Example 2-86
Configuration of Cisco Unified Communications Manager Express in SRST Mode 2-87
Phone Provisioning Options 2-89
Advantages of Cisco Unified Communications Manager Express in SRST Mode 2-90
Phone Registration Process 2-91
Configuring Cisco Unified Communications Manager Express in SRST Mode 2-92
Cisco Unified Communications Manager Express in SRST Mode Configuration Example 2-94
Summary 2-95
References 2-95
Module Summary 2-97
References 2-97
Module Self-Check 2-99
Module Self-Check Answer Key 2-102
Implementing Cisco Unified Communications Manager. Part 2 (CIPT2] v8.0 ©2010 Cisco Systems, Inc
Bandwidth Management and CAC Implementation 3-1
Overview 3-1
Objectives 3-3
Bandwidth Management Overview 3-4
Cisco Unified Communications Manager Codec Configuration 3-6
Review of Cisco Unified Communications l\ anager Codecs 3-7
Example: Codec Configuration 3-8
Local Conference Bridge implementation 3-10
Example: Implementing Local Conference I ridges at Two Sites 3-11
Transcoder Implementation 3-13
Example: Implementing a Transcoder at the Main Site 3-15
Configuration Procedure for Implementing ranscoders 3-17
Step 1: Add Transcoder Resource in Cisco Jnified Communications Manager 3-18
Step 2: Configure Transcoder Resource in isco IOS Software 3-19
Multicast MOH from Branch Router Flash Implementation 3-21
Multicast MOH from Branch Router Flash: I legion Considerations 3-23
Multicast MOH from Branch Router Flash: , .ddress and Port Considerations 3-24
Multicast MOH: Address and Port Incremer t Example 3-25
Example: Implementing Multicast MOH froi i Branch Router Flash 3-27
Configuration Procedure for Implementing lulticast MOH from Branch Router Flash 3-30
Step 1; Enable Multicast Routing on Cisco OS Routers 3-31
Step 2a: Configure MOH Audio Sources foi Multicast MOH 3-32
Step 2b: Configure Multicast MOH in Cisco|Unified Communications Manager 3-33
Step 2c: Enabling Multicast MOH at the ia Resource Groups 3-34
Step 3: Enable Multicast MOH from Branch!Router Flash at the Branch Router 3-35
Step 4a: Configure the Maximum Hops to Used for MOH RTP Packets 3-36
Step 4b: Use IP ACL at IP WAN Router Ints rface 3-37
Step 4c: Disable Multicast Routing on IP W VN Router Interface 3-38
Summary 3-39
References 3-39
Objectives 3-41
CAC Overview 3-42
CAC in Cisco Unified Communications Manjager 3-43
Standard Locations 3-44
Locations: Hub-and-Spoke Topology 3-45
Locations: Full-Mesh Topology 3-46
Configuration Procedure for Implementing Ijocations-Based CAC 3^17
Locations Configuration Example: Hub-and Spoke Topology 3-48
Step 1: Configure Locations 3-49
Step 2: Assign Locations to Devices 3-50
RSVP-Enabled Locations 3-51
Three Call Legs with RSVP-Enabled Locations 3-52
Characteristics of Phone-to-RSVP Agent Ci II Legs 3-53
Characteristics of RSVP Agent-to-RSVP Agent Call Leg 3-54
How RSVP Works 3-55
Configuration Procedure for Implementing RSVP-Enabled Locations-Based CAC 3-57
Example: RSVP-Enabled Locations Configuration 3-58
Step 1: Configure RSVP Service Parameters 3-59
Step 2: Configure RSVP Agents in Cisco IOS Software 3-63
Step 3: Add RSVP Agents to Cisco Unified Communications Manager 3-65
Step 4: Enable RSVP Between Location Pairs 3-66
Automated Alternate Routing 3-68
AAR Characteristics 3-69
AAR Example Without Local Route Groups and Globalized Numbers 3-70
AAR Example with Local Route Groups and Globalized Numbers 3-72
© 2010 Cisco Systems. Inc. Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
AAR Considerations 3.74
AAR Configuration Procedure 3-75
Step 1: Configure AAR Service Parameters 3-76
Step 3: Configure AAR Groups 3.77
Step 4: Configure Phones for AAR 3-78
SIP Preconditions 3-80
CAC Without SIP Preconditions 3-81
CAC with SIP Preconditions 3-82
SIP Preconditions Operation 3-83
SIP Preconditions Call Flow Summary 3-85
Fallback from End-to-End RSVP to Local RSVP 3-87
SIP Preconditions Configuration Procedure 3-89
Step 2a: Configure SIP Profile 3-90
Step 2b: Apply SIP Profile to Trunk 3-91
H.323 Gatekeeper CAC 3-92
Example: H.323 Gatekeeper Used for Call Routing (Address Resolution) Only 3-94
Using an H.323 Gatekeeper for CAC 3-96
Example: H.323 Gatekeeper Also Used for CAC 3-98
Providing PSTN Backup for Calls Rejected by CAC 3-100
Configuration Procedure for Implementing H.323 Gatekeeper-Controlled Trunks with CAC 3-102
Summary 3-104
References 3-104
Module Summary 3-105
References 3-105
Module Self-Check 3-107
Module Self-Check Answer Key 3-109
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 ©2010 Cisco Systems, Inc.
CIPT2
Course Introduction
Overview
Implemenling Cisco Unified Communicatioi s Manager, Part 2 (C1PT2) v8.0prepares you for
implementing a Cisco Unified Communications solution in a multisite environment. It covers
globalized call routing.Cisco Service Advertisement Framework (SAF) and Call Control
Discovery (CCD). tail-end hop-off (TEHO), Cisco Unified Survivable Remote Site Telephony
(SRST). and mobility features such as Devi e Mobility and Cisco Extension Mobility.
You will apply a dial plan for a multisite en ironment including TEHO, configure survivability
for remote sites during WAN failure and implement
i solutions toreduce bandwidth
requirements in the IP WAN. You will also mable Call Admission Control (CAC) including
Session Initiation Protocol (SIP) Preconditk ns and automated alternate routing (AAR).
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8 0 >2010 Cisco Systems. Inc
Course Goal and Objectives
This topic describes the course goal and objectives.
Course Flow
s implementation
[Course Introduction [ ; of Features and
. J Multisite Centra Iced Bandwidth 1 AppBcatena far
Aj MJtsite | Deployment CsS-P recessing Management j Multisite
M j Deployment ,. implementation Redundancy and CAC • Deployments
j Implementation , (Cot*.) Implements! on Implementation ; (Cont)
{Cont.) (Cor*) s
CCD
Lunch
Mufti sits Bandwidth
Deployment Management ana
Impiemen tiiofi CAC Implementation;
. MuKsite CCD
(Cont) 8andwkJtn (Cont) j
' ; Deployment (Cont)
M • fmplenientaliort Management Impiementation f
CentrafzerJ and CAC of Features and '
Cal-Processing Implementation Applications for
Redun&ncy Multisite
frnptementation Deplovmerts
The schedule reflectsthe recommended structure for this course. This structure allowsenough
time for the instructor to present the course information and for you to work through the lab
activities. The exact timing of the subjectmaterials and liibs depends on the pace of your
specific class.
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 ©2010 Cisco Systems. Inc
Additional References
This topic presents the Cisco icons and symbols thatareused in this course, as well as
information on where to find additional technical references.
Gatekeeper
Cisco Unrty _^^ Cisco Unified
Connection t^H Border Element
Gateway
Cisco Unified
Messaging Voice Router
Gas* a/ Personal
Communicator
Cisco Un ified
Coco Adapt ve Cisco Unified Communications
Secirty Appliance SRST Router Manager Express
Cisco Unified
SAF Enabled Communications
Router Manager Express with
Cisco Unity Express
You are encouraged to join the Cisco Certification Community, a discussion forum open to
anyone holdinga valid CiscoCareerCertification (such as CiscoCCIII",CCNA'. CCDA",
CCNP'. CCDP". CCIP". CCVP".or CCSP*). It provides a gathering place for Ciscocertified
professionals to share questions, suggestions, and information about Cisco Career Certification
programs and other certification-related topics. For more information, visit
http://\v\\\\.cisa>.eom;go certifications.
Implementing Cisco Unified Communications Manager,Part 2 (CIPT2) v8 0 © 2010 Cisco Systems, Inc.
Cisco Career Certifications: CCNP Voice
Recommended TrawwgThiouBtt
Cisco Leamw*g Partners
Course Introduction
i 2010 Cisco Systems, Inc.
8 Implementing Cisco Unified Communications Manager, Part2 (CIPT2) v8.0 ©2010Cisco Systems, Inc.
Module 1
Multisite Deployment
Implementation
Overview
In a multisite Cisco Unified Communications Manager deployment, special requirements exist
that are not necessary insingle-site deployments. To successfully deploy a multisite Cisco
Unified Communications Manager solution, you need to understand the issues and be aware of
their possible solutions.
This module discusses the issues in a multisite Cisco Unified Communications Manager
deployment, including selective public switched telephone network (PSTN) access and tail-end
hop-off (TEHO).
Module Objectives
Upon completing this module, you will be able to describe multisite deployment issues and
solutions, and describe and configure required dial plan elements.
This ability includes being able to meet these objectives:
• Explain issues pertaining to multisite deployment and relate the issues to multisite
connection options
• Describe solutions for multisite deployment issues
• Configure gateways and trunks in multisite environments
• Implement adial plan to support inbound and outbound PSTN dialing, site-code dialing,
and TL1IO in an international environment
1-2 Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 ©2010 Cisco Systems. Inc.
Lesson 1
Objectives
Upon completing this lesson, you will be able to explain issues pertaining to multisite
deployment and relate the issues to multisite connection options. This ability includes bein^
able to meet these objectives:
• Describe issues pertaining to multisite deployments
• Describe quality issues in multisite deployments
• Describe issues with bandwidth in multisite deployments
• Describe availability issues in multisite deployments
• Describe dial plan issues in multisite]dep!oyments
i
Cisco Unified
Communications
Manager
In a multisite deployment, several issues can arise. Of those issues, here are the most important:
• Quality issues: When real-time traffic like voice or video travels over a packet-switching
network such as an IP network, delay-sensitive packets have to be given priority to avoid
jitter resulting in decreased voice quality
• Bandwidth issues: Cisco Unified Communications solutions can include voice and video
streams, signaling traffic, management traffic, and application traffic, such as rich-media
conferences. The additional bandwidth that is required when deploying a Cisco Unified
Communications solution has to be calculated and provisioned. These tasks ensure thai
classical data applications and Cisco Unified Communications applications do not overload
the available bandwidth. You should optimize bandwidth consumption by eliminating
unnecessary IP WAN traffic.
• Availabilih issues: When you are deploying Cisco Unified Communications Manager
with centralized call processing. IP phones register with the Cisco Unified Communications
Manager over the IP WAN. If gateways in remote sites arc using the Media Gateway
Control Protocol (MGCP) as a signaling protocol, they also depend on the availability of
the Cisco Unified Communications Manager as a call agent. It is important to implement
fallback solutions for IP phones and gateways in scenarios in which the connection to Cisco
Unified Communications Manager is broken because of IP WAN failure.
• Dial plan issues: Directory' numbers are usually unique per site, but they can overlap
across multiple sites. Overlapping directory numbers and other issues,such as numbers that
are not consecutive, have to be solved by the design of a multisite dial plan. Various public
switched telephone network (PSTN)access codes in variouscountries arc anotherexample
of dial plan issues.
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) u8 0 © 2010 Cisco Systems. Inc.
• Network Address Translation (NAT) and security issues: Cisco Unified
Communications Manager and IP phones use IP primarily to communicate within the
enterprise. The use of private IP addresses is very common within the enterprise. When the
system should interact with a public IP network—for instance, when placing calls via an
Internet telephony service provider (ITSP)—then Cisco Unilied Communications Manager
and IP phone IP addresses have to be translated to public IP addresses. This translation
makes them visible on the Internet and, therefore, subject to attacks.
Note The issue of vulnerability when IP addresses are translated to public IP addresses is not
limited to multisite deployments.
Quality Issues
IP networks arc not designed to carry real-time traffic. Because of thenature of the network and
paeket-bv-packet delivery in which each packet could takea different path, there is no
guarantee that packets will arrive in thecorrect order at the destination. You can resolve this
issuebv using Real-Time Transport Protocol (RTP)sequence numbers.
Another issueis the fact that multiple usersand applications share the bandwidth, and the
actual required bandwidth varies significantly evenover short lapses of time. Therefore, the
bandwidth that is available for Cisco Unified Communications Manager traffic is
unpredictable. During peaks, packets need to bebuflered inqueues. Ifthecongestion occurs for
too long, buffers get tilled up and packets aredropped. Higher queuing delays and packet drops
aremore likely on highly loaded, slow links, such as WAN links that areused between sites in
a multisite environment. As a result, quality issues are common and need to be resolved by
implementing QoS. Otherwise, voice packets are subject tovariable delays (jitter) and packet
drops, bothof which impactvoice quality.
1-6 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v80 >2010 Cisco Systems, Inc
Quality Issues Example: Jitter and Packet Drops
The figure illustrates how packets are queued during congestion.
During peaks, packets cannot be sent immediately because ofinterface congestion, so they have
tobe stored in a buffer ("queued"). The time that the packet waits in such aqueue isreferred to
as the "queuing delay." The length ofthis delay can vary widely. Ifthe queue is full, newly
received packets cannot be buffered, so they get dropped (this action is called "tail drop").
Without any special treatment ofvoice packets, such as a FIFO processing model, the resulting
jitter and packet lossdecrease voice quality.
Bandwidth Issues
Implementing Cisco Unified Communications Manager. Part 2 (CIPT2) v8.0 ©2010 Cisco Systems, Inc.
Bandwidth Issues Example: VDice and Data Traffic Competing
for Bandwidth
The example illustrates the higher ovdVh ead ofvoice packets when comparing them with file
transfer packets.
Data packets:
• Large size
• Lower packet rate
• Small overhead
As shown mthe figure and as already mentioned, voice packets consume lots of bandwidth that
is caused by the overhead ofIP, UDP. and RTP headers that are added to small packets and
sent at ahigh packet rate. Data packets such as afile transfer also add 40 bytes ofoverhead (20
bytes IP and 20 bytes TCP), bu, the payload is as large as possible (filling up the maximum
transrmsston umt. or MTU^ypically about 1500 bytes. Because of the large payload the
packet rate ,s also lower, and overhead is not added, as is often the case with voice packets.
Because ofthe inefficiency of voice packets, all unnecessary voice streams should be kept
away from .he IP WAN. Media resources, in particular, can be optimized in such away that
they do not have to cross the IP WAN all the time, thus conserving valuable bandwidth You
can achieve this optimization by utilizing local media resources.
Cisco Unified
Communications
Manager
in the example, aconference bridge has been deployed at the mam site. 1here >s no conference
bridge at the remote site. If remote IP phones join aconference, their Rl Pstreams are sent
across the WAN to the conference bridge. The conference bridge mixes the received audio
streams and then sends them back again to the IP phones over the IP WAN.
There arc three members in the conference in this example, and all of .hem are physically
locked at the remote site. In total, three RTP streams are flowing toward the coherence bndge.
and three RTP streams are flowing back to the remote site. Assuming ^f^ett.ngs each
RTP stream requires 80 kb/s (ignoring the Layer 2overhead), resulting in 240 kb/s of IP WAN
bandwidth that is required bv this voice conference. If the conference bndge was not located n
life olr side ofthe IP WAN. this traffic would have avoided the WAN link entirely, since all
participants ofthe conference are local to the remote site.
Availability Issues
Remote Cluster
Cisco Unified
Communications
Manager
In the example, there is a main site with an inlerclusler trunk to a remote Cisco Unified
Communications Manager cluster. There is also aremote site with IP phones that register at the
Cisco Unified Communications Manager cluster that is located at the main site. ASIP trunk is
used to connect to an ITS!'.
Ifa WAN failure occurs, no calls to the other cluster orto the ITSP are possible. In addition, all
phones that arelocated at the remote site lose registration with Cisco Unified Communications
Manager, sothev do not operate atall. They cannot even place calls within the remote site.
Note Adeployment as shown inthe exampleis badly designedbecause of the lack of IP WAN
backup.
1-12 ImplementingCisco Unified Communications Manager. Part 2 (CIPT2]v8.0 © 2010 Cisco Systems, Inc
Dial Plan Issues
This topic describes dial plan issues in Cisco Unified Communications Manager multisite
environments.
In a multisite deployment, dial plan design requires the consideration of several issues that do
not exist in single-site deployments:
• Direct inward dialing (DID) ranges and E.164 addressing: When you are considering
integration with the PSTN, internally used directory numbers have to be related to external
PSTN numbers (E.164 addressing). Depending on the numbering plan (fixed or variable)
and services that are provided bv the PSTN, these solutions are common:
Each internal directory number relates to a fixed-length PSTN number: In this
case, each internal director; number has its own, dedicated PSTN number. The
directory number can, but does not have to, match the least significant digits of the
PSTN number. In countries with a fixed-numbering plan, such as the North
American Numbering Plan (NANP). the four-digit station codes, for instance, are
used as internal director;' numbers. If these numbers are not unique, digits of office
codes or administratively assigned site codes might be added, resulting in five or
more digits being used for internal directory numbers.
Another solution is not to reuse any digits of the PSTN number, but to simply map
each internally used director.' number lo any PSTN number that is assigned to the
company. In this case, the internal and external numbers do not have anything in
common.
1-14 Imolementing Cisco Unified Communications Manager. Part 2 (CIPT2) v8.0 ©2010 Cisco Systems. Inc
• Optimized call routing: An IP WAN between sites with PSTN access allows PSTN toll
bypass. Cisco Unified Communications Manager servers route calls between sites over the
IP WAN instead of using the PSTN (toll bypass). In such scenarios, the PSTN should be
used as a backup path only in the case of WAN failure. Another solution, which extends the
idea of toll bypass,is to use the IP WAN also for PSTN calls: With tail-endhop-off
(THHO), the IP WAN is used as much as possible, and the gateway that is closest to the
dialed PSTN destination is used for the PSTN breakout. Finally, a backup path over the
PSTN should be enabled for when a call cannot be sent over the IP WAN (for example, if
the IP WAN is down or the maximum number of allowed calls is reached).
• Various PSTN requirements: Various countries—and sometimes even various PSTN
providers withinthe same country—can have variousrequirements regarding the PSTN
dial rules. This situation can cause issueswhen calls can be routedvia multiplegateways,
For example, if the requirements of a primary gateway are different from the requirements
of a backup gateway, numbers have to be transformed accordingly.
p,„ The calling number (the Automatic Number Identification, or ANI) ofcalls that are being
^ received from the PSTN can be represented in various ways: as a7-digit subscriber
number, as a 10-digitnumber including the area code, or in international format with the
country code in front of the area code. To standardize the calling number for all calls, the
format that is used mustbe known, and the number has to be transformed accordingly. In
countries where PSTN numbers do not have fixed lengths, it is impossible to detect the type
(local, national, or international) of the number by the number only by looking at its length.
Insuch cases, thetype of number has to be specified in signaling messages (forexample,
by the ISDN type of number, or TON).
• Scalability: In large or very large deployments, dialplanscalability issues arise. When
interconnecting multiple Cisco Unified Communications Manager clusters or CiscoUnified
Communications Manager Express routers viatrunks, it is difficult to implement a dial plan
on an any-to-any basis where each device or cluster needs to know which numbers or
prefixes are found at which other system. In addition to the need to enter almost the same
dial plan ateach system, a static configuration does not reflect true reachability. Ifthere are
any changes, the dial plans ateach system have to be updated. Although there aresolutions
thatallow centralized dial plan configuration (forexample, H.323 gatekeepers), in very
large deployments a dynamic discovery ofdirectory number ranges and prefixes would
simplify the implementation and provide a more scalable solution.
NANP.
No DID Variable-Length E.164
Aulo-Attendanl Used Addressing with DID
Cisco Unified
Communications
Manager
"fhe example features a main site in the United Slates. The NANP PSTN number is408 555-
1234. DID isnotused. All calls placed to themain site are managed by anattendant. There is a
remote site in Gennanv with a PSf N numberof 149 404 13267. The Cierman location uses
four-digit extensions, and DID isallowed, since digits can be added to the PSTN number.
When calling the German office attendant (not knowing aspecific extension), users in the
United States would dial +9 011 49 404 13267. If they know that they want to contact
extension 1001 dircclK. thev would dial+9 011 49 404 13267 1001.
Examples:
• Witriir It S : "9 1 408 555-1234" or '1 555-1234" twilrifi Bia same area code)
• U.S to Austria1 "9 011 43 1 12345"
• WitHii Austria: "0 0 1 12345" or "0 1234" (withn the same area code)
• Austria to U.S.. "0 00 1 406 555-1234 " (1 is country code, not national access code)
, _ „ _, VsriSUle-Length Numbing
Components £|*ea Numbering Plan " *
Country code 1 43
Area code 3 digit 1-4 (Halt*
3-tfio.texchsng» cads ♦ 3«mor»-<Igflt
Subscriber numoer
^cfigistaawtcoas
PSTN access code 9 "~ 0
National access code 1 0
00 or*
International access code 011
(•+' uwd by e#Jpfion«)
A fixed numbering plan, such as the plan used in North America, features fixed-length area
codes and local numbers. An open numbering plan, such as the plan used in various countries
that have not yet standardized their numbering plans, features variance in the length of the area
code or the local number, or both.
A country code is used to reach the particular telephone system for each country or special
service.
An area code is used within many countries to route calls to a particular city, region, or special
service. Depending on the country or region, it may also be referred to as a Numbering Plan
Area (NPA), subscriber trunk dialing code, national destination code, or routing code.
The subscriber number represents the specific telephone number to be dialed, but does not
include the country code, area code (if applicable), international prefix, or trunk prefix.
A trunk prefix refers to the initial digits to be dialed in a call within the United States,
preceding the area code and the subscriber number.
An international prefix is the code to be dialed before an international number (the country
code, the area code if any, and then the subscriber number).
The table contraststhe NANP and a variable-length numbering plan (the Austrian numbering
plan, in this example).
• Use of # key
Different implementation in Cisco IOS Software (simple) versus
Cisco Unified Communications Manager (complex)
- Convenient
From an implementation perspective, the simplest way to detect end of dialing is to wait for an
interdigit timeout to expire.This approach, however, provides the least comfort to the end user
because it adds postdial delay. In an environment with only few numbers of variable length (for
example, the NANP. where only international calls are of variable length), waiting for the
interdigit timeout ma\ be acceptable. However, even in such an environment, it may make
sense to at least reduce the value of the timer, because the default value in Cisco Unified
Communications Manager is rather high (15 s).
Note In Cisco Unified Communications Manager, the interdigit timer is set by the dusterwide
Cisco CallManager service parameter T302 timer that is found under Device > General.
In Cisco IOS Software, the default for the inlerdigit timeout is 10 seconds. You can modifj this
value using the voice port timeouts interdigit command.
Another solution for detecting end of dialing on variable-length numbers is the use of the # ke\.
An end usercan press the # kej to indicate thatdialing hasfinished. The implementation of the
# kev is different in Cisco Unified Communications Manager versus Cisco IOS Software. In
Cisco IOS gatewa\s. the # is seen as an instruction to stop digit collection. It is not seen as part
of the dialed string. Therefore, the # is not part of the configured destination pattern. In Cisco
Unified Communications Manager, the # is considered to be part of the dialed number, and
therefore its usage hasto be explicitly permitted by the administrator by creating patterns that
include the it. If a pattern includes theit. the # hasto be used; ifa pattern doesnot include the#.
the pattern is not matched if the user pressed the # key. Therefore, it is common in Cisco
Unified Communications Manager to createa variable-length patterntwice: once with the it at
the end and once without the #.
1-18 Implementing Cisco Unified Communications Manager. Part 2 (CIPT2) v8 0 ©2010 Cisco Systems, Inc
An alternative wayof configuring such patterns is to end the pattern with ![0-9#]. In thiscase,
a single pattern supports bothwaysof dialing: withthe # and without the #. However, be aware
ff^ that the use ofsuch patterns can introduce other issues. For example, when using discard digits
instructions that includeTrailing-^(for example, PreDot-Trailing-#). This discard digit
instruction will have an effectonly when there is a trailing # in the dialed number. If the # was
p.,* not used, the discard digit instruction is ignored and hence the PreDot component ofthe discard
IM digit instruction is also not performed.
Allowing the use of the # to indicate end of dialing providesmore comfortto end users than
having them wait forthe interdigit timeout. However, thispossibility has to be communicated
to the end users, and it shouldbe implemented consistently. As mentioned earlier, it is
automatically permitted in Cisco IOS Softwarebut not in Cisco UnifiedCommunications
Manager.
The third way to indicate end of dialing is the use of overlapsend and overlapreceive.If
overlap is supported end-to-end, the digits that are dialed by the end user are sent one by one
over the signaling path. Then the receiving end system can inform the calling device once it has
received enough digits to route the call (number complete). Overlap send and receive is
common in some European countries such as Germany and Austria. From a dial plan
implementation perspective, overlap send and receive is difficult to implement when different
PSTN callingprivileges are desired. In this case, you have to collectenoughdigits locally(for
jte example, in Cisco Unified Communications Manager or Cisco IOS Software) to be able to
decide to permit or deny the call. Only then can you start passing digits on to the PSTN one by
one using overlap. For the end user, however, overlap send and receive is very comfortable,
pw because each call is processed as soon asenough digits have been dialed. Thenumber of digits
§^ that are sufficient varies perdialed PSTN number. Forexample, one local PSTN destination
may be reachable by a seven-digit number, whereas another local number may be uniquely
identified only after receiving nine digits.
L
Optimized Call Routing and PSTN Backup
Using an 1P WAN enables savings on the cost of PSfN calls in a multisite en\ ironment.
"there are two wa\s to save costs for PSTN calls in a multisite deployment:
• Toll bypass: Calls between sites that use the IP WAN instead of the PSTN are toll-bypass
calls. The PSTN is used only when calls over the IP WAN are not possible—either because
of a WAN failure or because the call is not admitted by Call Admission Control (CAC).
• TEHO: TEHO extends the concept of toll bypass by also using the IP WAN for calls to the
remote destinations in the PS'fN. With TIT 10. the IP WAN is used as much as possible and
PSTN breakout occurs at the gateway that is located closest to the dialed PSTN destination.
Local PSTN breakout is used as a backup in case of an IP WAN or CAC failure.
Caution Some countries do not allow the use of TEHO. When implementing TEHO, ensure that the
deployment complies with legal requirements.
When using the IP WAN lo reach remote PS'fN destinations or internal director) numbers at a
different site, it is important to consider backup paths. When the IP WAN is down or when not
enough bandwidth is available for an additional voice call, calls should be routed via the local
PSTN gateways as a backup path.
1-20 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2)v8 0 ©2010 Cisco Systems. Inc
Example: TEHO
The figure illustrates the use of TEHO in a multisite deployment.
Example: TEHO
l^.
403-555-6666
In the example, a call from Chicago to San Jose, California, would berouted inthe following
way:
1. A Chicago userdials9 1 408 555-6666, the number fora PSTN phone thatis located in
San Jose.
IM
Cisco Unified
Communications
Manager
1001-1099 •* ••1001-1099
Nonconsecutive
2000-2157 +• -+• 2158-2364
Numbers
2365-2999
Inthe example. IP phones ai the main site use directorv' numbers 1001 to 1099. 2000 to 2157.
and 2365 to 2999. At the remote site. 1001 to 1099 and 2158 to 2364 are used. There are two
issueswith thesedirector, numbers: 1001 to 1099overlap; and thesedirectory numbers exist at
both sites, so they are notunique throughout the complete deployment. Inaddition, the
nonconsecutive use ofthe range 2000 to 2999 (some numbers outof this range areused at the
remote site, and the others are used at the main site) wouldrequire lots of entries in call-routing
tables, since the ranges can hardly be summarized by one (ora few) entries.
Note The solutions to the problems that are listed inthis lesson are discussed inmoredetail inthe
next lesson of this module.
1-22 Implementing Cisco Unified Communications Manager, Part2 (CIPT2) v80 © 2010 Cisco Systems, Inc.
ttM.
One of the issues in international deployments is various PSTN dial rules. For example, in the
United States, the PSTN access code is 9, while in most countries in Europe, 0 is used as the
PSTN code. The national access code in the United States is I, while 0 is commonly used in
Europe. The international access code is 011 in the United States,while 00 is used in many
European countries. Some PSTN provider networks require the use of the ISDN TON, while
others do not support it. Some networks allow national or international access codes to be
combined with ISDN TON. Others require you to send the actual number only (that is, without
any access codes) when setting the ISDN TON.
Thesame principle applies to the calling-party number. As mentioned earlier, in variable-length
numbering plans, the TON cannot be detectedby its length.Therefore, the only way to
determine whetherthe receivedcall is a local,national, or international call is by relying on the
availability of the TON information in the receivedsignalingmessage.
**»
Some countries thathavevariable-length numbering plansuseoverlap sendandoverlap
receive. With overlap send, a numberthat is dialed by an end user is passedon to the PSTN
digit by digit. Then the PSTN willindicate when it has received enough numbers to routethe
call. Overlap receivedescribes the same conceptin the oppositedirection:when a call is
received from the PSTN in overlap mode, the dialed number is delivered digit by digit, andnot
en bloc. Some providers thatuse overlap sendtoward theircustomers do notsend the prefix
that is configured forthe customer trunk, but only the additional digitsthat are dialed by the
user who initiates the call.
When dialing PSTN numbers in E.164 format (that is,numbers thatstart with the country
code), the + sign is commonly prefixed to indicate that the number is in E.164 format. The
advantage of using the+ sign as a prefix forinternational numbers is thatit is commonly
known as a + sign around the world. Incontrast, PSTN access codes suchas 011 (used in the
NANP) or 00 (often used in Europe) areknown onlyin the respective countries.
L
Finally, emergency dialing can be an issue in international deployments. As various countries
have various emergenc\ numbers and various ways to place emergency calls, users are not sure
how to dial the emergency number when roaming to other countries. An international
deployment should allow roaming users to use their home dialing rules when placing
emergency calls. The system should then modify the called number as required at the respective
site.
1-24 Implementing Cisco Unified Communications Manager. Part 2(CIPT2) v8.0 ©2010 Cisco Systems, Inc.
Issues Caused by Different PSTN Dialing
Different local PSTN dial rules can cause several issues, especially in international
deployments.
The main problem that needs to be solved in international environments is how telephone
numbers ofcontacts should bestored. Address book entries, speed and fast dials, call list
entries, and other numbers should be in a format that allows them tobe used atany site,
regardless of the local dial rules that apply to the site where the user is currently located.
The same principle applies to numbers that are configured by the administrator—for example,
the target PSTN number for automated alternate routing (AAR) targets. Call-forwarding
destinations should also be in a universal format—a format that allows the configured number
to be used atany site. The main reason for auniversal format is that amultisite deployment has
several features'that make itdifficult to predict which gateway will be used for the call. For
example, aroaming user may use Cisco Extension Mobility or Device Mobility. Both features
allow an end user toutilize local PSTN gateways while roaming. Ifno universal format isused
to store speed dials or address book entries, itwill be difficult for the end user to place aPSTN
call to a number that was stored according to the NANP dial rules while incountries that
require different dial rules. Even when not roaming, the end user can use TEHO orLeast Cost
Routing (LCR). so that calls break out to the PSTN at aremote gateway, not at the local
gateway. Ifthe IP WAN link to the remote gateway is down, the local gateway is usually used
as abackup. How should the number that isused for call routing look in such an environment?
It isclearly entered according to local dial rules by the end user, but ideally itischanged to a
universal format before call routing is performed. Once thecall has been routed and the egress
gateway has been selected, the number could then be changed as required by the egress
gateway.
The main scalability issue oflarge deployments is that each call routing domain (for example, a
Cisco Unified Communications Manager cluster or aCisco Unified Communications Manager
Express router) needs to be aware of how to getto all other domains.
Such adial plan can become very large and complex, especially when multiple paths (for
example, a backup path for I'EHO) have to be made available. As each call routing domain has
to be aware ofthe complete dial plan, astatic configuration does not scale. For example, anv
changes in the dial plan have to be applied individually at each call routing domain.
Centralized H.323 gatekeepers orSIP network services can be used to simplify ihe
implementation ofsuch dial plans, because there is no need toimplement the complete dial
plan at each call routing domain. Instead ofan any-to-any dial-plan configuration, only ihe
centralized component has tobe aware ofwhere lo find which number. This approach,
however, means thai vou rely on a centralized service. Ifthe individual call-routing entities
have noconnectivity to the centralized call-routing intelligence, all calls would fail. Further, the
configuration isstill static. Any changes atone call-routing domain (for example, new PSTN
prefixes because of changing the PS'fN provider) have to be implemented also at the central
call-routing component.
In addition, these centralized call-routing services do not have built-in redundancy.
Redundancv can be provided, but requires additional hardware, additional configuration, and so
on. Redundancv is not an integrated part of the solution.
1-26 ImplementingCisco Unified Communications Manager. Part 2 (CIPT2]vB.O ©2010 Cisco Systems, Inc.
Answer Key
The correct answers and expected solutions for the activities that are described in this guide
appear here.
Implementing Cisco Unrfied Communications Manager, Part 2(CIPT2) v8.0 ©2010 Cisco Systems, Inc
router eigrp SAF
service-family ipv4 autonomous-system 2
shutdown
Step 13 Verify that the IP path to all learned patterns is marked unreachable by using the
show voice saf dndb all command.
Activity Verification
You have completed this task when you attain these results:
• The BR-.t router learns patterns by CCD as described in the activity procedure.
. Verifv that calls can be placed to patterns learned by CCD while being in SRST and MGCP
fallback mode.
Step 14 Place calls from your BR-.v site to all other three sites.
• Verifv the path that each call takes by using the debug isdn q931 command at
all four gateways.
• Use site code dialing for calls to the other pod: 8-5 lv-200! or 8-5I r-2002 to
IIQ-v phones and 8-52.V-3001 to the BR-v phone.
• Use four-digit extensions for calls to the local HQ-.t phones: 2001 or 2002.
Note When acall is placed to aphone located at one of the BR sites, the SIP INVITE is sent to
the advertising device (CUCM1-X or CUCM1-y). As the BR sites are in SRST mode, Cisco
Unified Communications Managers will route the calls to their BR gateway via the PSTN
basedon the existing CFUR configuration.
Lab Guide 95
© 2010 Cisco Systems. Inc.
Note This dial peer is required for the CCD PSTN backup calls. The learned toDID rule changes
the internally used site-code numbers from 8 followed by seven digits to E.164 numbers with
a plus prefix The CCD PSTN backup calls then match this dial peer where the + is stripped
and then 011 is prefixed.
The learned pattern 851x-2XXX refers to the HQ-x site. Until now calls between the two
sites of the same pod used four-digit dialing. CCD cannot advertise the HQ-x site with 4
digits to the BR-x SAF client and with site code prefixes to other SAF clients (the SAF clients
of the other pod). Usually, when using CCD, the internally used pattern for a given site has
to be the same at all sites.
However, the problem can also be solved in a way that allows users to continue using 4
digits for intersite calls within the same pod. You need to modify the dialed four- digit number
2XXX at the BR-x router to 851x2XXX before the outbound dial peer is selected.
Step 10 On the BR-.rrouter configure the following number expansion in order to allow
BR-.v users to continue using four-digit intersite dialing towards the HQ site of the
local pod (HQ-.t):
Note Byconfiguring the number expansion command calls to four-digit numbers starting with 2
(for example 2001) are expanded to site code dialing format (851x2001) before the selection
of the outgoing dial peer. The expanded number is used to select the outgoing dial peer
(dial-peer 8 in this case) which refers to SAF-learned patterns. The BR-xgateway finds a
match in a learned pattern (851x2XXX) that is currently marked unavailable. Therefore, a
PSTN backup call is placed using the learned toDID rule (4:+5551x555). The resulting call to
•••5551x5552001 matches dial peer 999, which sends the call to the PSTN with a called
number of 0115551x5552001.
Note Use the interface that connects the HQ-x route with the BR-x router.
Step12 Break the SAT connection between your HQ router (HQ-.v) and your BR router
(BR-.r) in order to have the IP palhs marked unreachable by enleritig the following
commands in global configuration mode of the \K)-x router:
94 Implementing Cisco UnifieO Communications Manager, Part 2 (CIPT2) v8.D ®2010 Cisco Systems. Inc.
-Jfe
Note The learned pattern 852x-3XXX refers to the BR-x site itself.This pattern will not be used at
a phone that is located at the BR-x site because four-digit dialing is used for internal calls. If
it was dialed, the call cannot use the advertised IP path (to Cisco Unified Communications
Manager CUCM1-x) because the IP WAN link is down. Remember that the BR-x gateway
normallydoes not use a local dial plan as it is configured as an MGCP gateway. It will only
look to its local call routing table when the connection to the HQ site is broken, if a BR-x
user dials a number out of the 8-52x-3XXX range during SRST mode, the BR-x gateway
finds a learned pattern that is currently marked unavailable Therefore, the PSTN backup
path (toDID 4:+6652x555) is used to place a PSTN backup call. When this call is set up by
the BR-xgateway, the PSTN routes the call back to the BR-xgateway. This means that
calling to the own site by site-code dialing works, but it would use two ISDN circuits as the
call is hairpinned at the PSTN
Enable Call Routing for CCD-learned Patterns and for CCD PSTN Backup Calls
Step9 Onthe BR-\ router configure the following dial peersin orderto enable the gatewav
to use CCD-learned patterns and to enable CCD PSTN backupcalls:
Note Thisdial peer instructs Cisco Unified Communications Manager Expressto look to the CCD-
learned patterns when a user diats 8 followed by seven digits.
The learned patterns 851y-2XXX and 852y-3XXX refer to the other pod. The IP destination
for both patterns is the Cisco Unified Communications Manager of the other pod
(CUCM1-y) However, as the IP WAN is down, the PSTN backup path has to be used
Based on the learned toDID rules (4:+5551y555 for pattern 851y-2XXX and 4:+6652y555 for
pattern 852y-3XXX) the BR-x gateway will placea directcall to the respective gatewayof
the other pod (HQ-yor BR-y).
Lab Guide
) 2010 Cisco Systems. Inc
exit-service-family
i
Step 6 Configure outbound dial peers for intersite calls to use SAP:
dial-peer voice 8 voip
destination-pattern 8
session target saf
Caution The X in the pattern of the show voice saf dndb detailcommand is case sensitive.
Note In each pod, Cisco Unified Communications Manager is the call agentthatadvertises the
HQ-x and BR-x patterns. Therefore, at the BR-x router 851x-2XXX and 852X-3XXX should
be listed as reachable bySIPat 10.x 1.1 while 851y-2XXX and 852y-3XXX should be listed
as reachable by SIP at 10.y.1.1.
The BR-x router will never use thelearned SIP path. As long as there isnoIPconnectivity
problem, the ISDN PRl is MGCP-controlled and the BR-x phones are registered to Cisco
Unified Communications Manager. Therefore no call routing occurs at the BR-x gateway
under normal situation.
The learned patterns are only used incase ofIPWAN failure. In this case, however, the IP
path ofthe learned patterns ismarked unavailable andthe BR-x gateway which then
operates in SRSTand MGCP fallback mode will use the learned patterns to placeCCD
PSTN backup calls based on the learned toDID rules.
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 ©2010 Cisco Systems, Inc.
Step 10 Repeat the test calls. Verify thatthe callsarererouted via the local PS'fN gatewa;
mm b\ usingdebug isdn q931 command at bothgateways.
Note Be aware of the call flow in this scenario. After calling a phone of the other pod using site-
code dialing a CCD-learnedpattern is matched. The SAF-enabledSIP trunk is not useO
because the IP path is marked unreachable (dueto shutting down SAF). The ToDID rule is
*•• applied to the matched pattern and a call to the {now globalized) number is placed using the
AAR CSS.
The call matches the intersite pattern, which refers to the non-SAF-enabled SIP trunk as first
option and to the local routegroupas second option. As ihe first option does notwork (due
"•* tothe access list), the call is finally sent via the PSTN using the local gateway.
imm Step 11 Re-enable the SAF connection between the two HQ routers by entering the
following commands in global configuration mode at your HQ~.v router:
<**
Step 12 Remove the access listfrom the serial interface at yourIIQ-.v router.
Lab Guide
& 2010 Cisco Systems, Inc
(CUCMl-v) by reapplying the access-list that was already used in earlier lab
exercises in global configuration mode at your HQ-x router:
interface serial .. .
ip access-group 100 in
Note Use the interface that connects the HQ-x route with the HQ-y router.
Caution IP connectivity between the two Cisco Unified Communications Managers needs to be
broken in order to avoid that the CCD PSTN backup call is sent over the (non-SAF-enabled)
SIP trunk that has been created in an earlier lab exercise. Remember that intersite calls
placed to the other pod are not sent via the PSTN but via the IP WAN. The same applies to
TEHO calls placed to PSTN destinations attached to the HQ and BR gateways that are
located in the other pod. By applying the access list the SIP trunk is not operational anymore
and intersite calls placed to the other pod will use the PSTN as a backup path.
Make sure that the access list is applied at the HQ router of both pods. If the access list is
only applied at your local HQ router, you will experience very high post-dial delays when you
try to reach the other pod.
The reason for the post-dial delay is that the originating Cisco Unified Communications
Manager has to wait for a timeout when not being notified that the packet has been dropped.
Ifthe HQ router of the other pod does not drop the SIP INVITE sent by the local Cisco
Unified Communications Manager, then only the response packet that is originated by the
Cisco Unified Communications Manager of the other pod is dropped inbound at your HQ
router In this case only the Cisco Unified Communications Manager of the other pod is
notified that its response packet was dropped. The local, originating Cisco Unified
Communications Manager is not aware of any packet drops and hence has to wait for the
timeout to expire. After timeout expiration it willretry the call setup and the timeout will have
to be waited for again.
When both HQ routers are configured with the inbound access list at their interconnecting
serial interface, then Cisco Unified Communications Manager will always be immediately
aware of the network issue (packet drop) and switch over to the PSTN backup path without
additional delay.
• Break the SAF connection between your HQ router (I IQ-.v) and the IIQ router of
the other pod (HQ-v) by enteringthe following commands in global
configuration mode at your HQ-x router:
Step 9 Use Cisco Unitied RTMT to verity that the IP paths of the learned patternsare
marked unreachable.
implementing Cisco Unified Communications Manager, Part2 (CIPT2) u8.0 ©2010 Cisco Systems, Inc.
Step 32 Move the SAFSlPTrunk to Selected SAF Trunks.
Step 33 Check the Activated Feature check box and click Save.
Activity Verification
You have completed this task when you attain these results:
• Verify registration of the external SAF client.
Step 1 from the I IQ-.v router, enter the show eigrp service-family ipv4 clients command
in privilege mode to verify that Cisco Unified Communications Manager has
registered with the SAF forwarder.
• Verify the learned patterns by using RTMT.
Step 2 Install and launch Cisco Unified Real Time Monitoring Tool (RTMT) on your
student PC.
Note You should see a pattern of 851/2XXX with a toDIDof 4:+5551y555 and a pattern of
4M
852/3XXX with a toDIDof4:+6652y555 Bothpatterns should be reachable by SIP at IP
address I0.y.1.1.
Note Cisco Unified Communications Manager will set up a call to the learned IP address using the
learned protocol (SIP inthiscase). The receiving Cisco Unified Communications Manager
cluster will strip the called numberto the internally used four-digit directory numbers
because the SAFSlPTrunk was configured with significant digits 4.
Step 12 Configure the trunk with the following settings, and then click Save.
• Device Name: SAFSlPTrunk
• Device Pool: Trunks
• Significant Digits: 4
• CallingSearchSpace: Trunk_css.
• SIPTrunk Security Profile: Non Secure SIP Trunk Profile
• SIP Profile: Standard SIP Profile
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 ©20)0 Cisco Systems, Inc.
• Once the configuration ofthe other pod has finished, enter the show eigrp service-family
ipv4 2 neighbors command at your HQ-.v and BR-.r router:
At the HQ-x router you should see the BR-.r router and the HQ-y router as
neighbors.
Atthe BR-.\ router you should seethe IIQ-.v router as neighbor.
Step 5 Configure the SAF security profile with the following settings and then click Save.
• Name: HQ_SAF_Profile
• Username: SAFl'SER
• Password: SAFPASSWORD
Note The username and password need to match the credentials configured in Task 1
Step 6 Navigate to Advanced Features >SAF >SAF Forwarder and click Add New.
Step 7 Configure the SAF forwarder with the following settings:
• Name: HQ.v_SAF
• Client Label: HQr_SAF
Note The client label needs to match the external client configured at Task 1
Lab Guide 87
© 2010 Cisco Systems, Inc
Task 1: Configure SAF Forwarder Functionality on the HQ-x
and BR-x Router
In this task, you will configure the HQ-.r router to act as an SAF forwarder.
Activity Procedure
Complete these steps:
Step 1 Open a Telnet session to your HQ-.r router(IOjc. 1.10I).
Step 2 Enter theconfigure terminal command to access theglobal configuration mode.
Step 3 Enter the following commands:
i
topology base
external-client HQx_SAF
exit-sf-topology
exit service-family
password SAFPASSWORD
Note Do not use special characters like spacesor dashesfor theexternal client definition.
topology base
exit-sf-topology
neighbor 10.x.250.101 LoopbackO remote 16
exit-service-family
Activity Verification
You have completed this task when you attain these results:
• The SAF forwarders have been configured as described in the activity procedure.
86 Implementing Cisco Unified Communications Manager. Part 2(CIPT2) v8.0 ©2010 Cisco Systems. Inc.
Lab 5-1: Implementing Cisco SAF and CCD
Complete this lab activity lopractice what you learned inthe related module.
Activity Objective
In this activity. \ou will implement CCD using Cisco SAF clients and forwarders. After
completing this activity, you will be able to meet these objectives:
• Configure SAF forwarder functionality on the HQ-.t and BR-.r routers
• Contigurc Cisco Unified Communications Manager asan advertising and requesting SAF
client
• Configure Cisco Unified Communications Manager Express onthe BR-.r router asa
requesting SAF client
Visual Objective
The figure illustrates what you will accomplish inthis activity.
Phnnel-* Phane2-'
DHCP 1
4 ICx 30/24 ioy_;y
101
Enable CCD to
leam and advertise
call routing
information
"' Frame S^SS PSTN
J2£
j&
Required Resources
These are the resources and equipment that arc required tocomplete this activitv:
Cisco Unified Communications Manager
Student PC
Cisco IP Phones
H.323 galeway
PSTN with PSTN phone
Lab Guide 85
© 2010 Cisco Systems. Inc.
Step 15 From the Selecta Servicedrop-down menu, choose the EM service. Click Next.
Step 16 Click Subscribe, "fhe Cisco F.xiension Mobility service isdisplayed under
Subscribed Services.
Step 18 Click Reset in the Phone Configuration window to reset the phone.
Step 19 Repeat the previous steps (enabling Cisco Unified Communications Manager
Fxtension Mobility and subscribing tothe Cisco Extension Mobility IP phone
service) for Phone2-.r and Phone.Vr.
Note As an alternative loperforming steps 3to19 you could have activated the Enterprise
Subscription check box when configuring theCisco Extension Mobility IP phone service.
Enterprise subscriptions apply to all phones and to all device profiles.
Activity Verification
You have completed this task whenyou attain these results:
• You can log in and log out at Phone I-x, Phone2-jr, and Phone3-.r by performing Ihe
following steps:
Step 1 Press the Services button.
Step 4 The phone will reset and should (hen be loaded with your device profile, fhe
director, numbershouldchangeto 2405.
Step 5 Place calls to internal and external (PSTN) destination.
Note Cisco Extension Mobility does not modify device level settings such as region and location
ordevice CSS and AAR CSS. These parameters arenot configurable in the device profile.
The line CSS of the phone where a Cisco Extension Mobility user logs misupdated with the
line CSS of the device profile. In this lab. the line CSS isHQ_css. This CSS provides access
to theHQ translation patterns. Therefore, the PSTN dial rules ofthe HQ sitehave to be
used.
• The configured service parameters are working. This can be verified by performing the
following steps:
Step 1 Log in at Phone3-.v and place acall. Then wait for the maximum login timer (3
minutes) to expire. You should be automatically logged out when the timer expires.
Step 2 Log in again at Phone3,v. Verify that the call list was cleared alter logout: use the
Redial softkey and verify that the phone does not remember Ihe last destination.
Step 3 Do not log out. Log in at Phone 1-.t before the 3-minute timer expires. Once you
have logged in at Phone I-.t. you should be automatically logged out at Phone3-.v.
because the multiple login behavior has been set lo auto-logout.
Note After logging out or being logged out of a phone, the phone reconfigures itself to its standard
settings
84 Implementing Cisco Unified Communications Manager. Part 2(CIPT2) v8.0 ©2010 Cisco Systems, Inc.
Task 4: Add the Cisco Extension Mobility IP Phone Service and
Subscribe to IP Phones and Device Profiles
Inthistask, vou will addthe Cisco Fxtension Mobility IP phone service andsubscribe it to IP
phones and the device profile.
Activity Procedure
Complete these sleps:
• ASCII Service: EM
Note After you click Save, the Parameters pane will appear—Cisco Extension Mobility does not
need any additional parameters to be specified.
Step 6 In User Management > End User, verify that the end user "andy" has been added to
Cisco Unified Communications Manager.
Activity Verification
You have completed thistaskwhen you attain these results:
• The end user "andv" is configured in User Management > End User as described in the
activity procedure.
• The device profile andy_dp is assigned to the end user as described in the activity
procedure.
82 Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 ©2010 Cisco Systems, Inc
Task 2: Create a Device Profile for a User
In this task, vou will create a device profile for the user of Phone l-.v.
Activity Procedure
Complete these steps:
Step 1 Nav igate to Device >Device Settings >Device Profile.
Step 2 Click Add New.
Step 3 Tor ihe phone model, choose Cisco 7965. Click Next.
Step 4 Keep the SCCP signaling option and click Next.
Note The phone button template depends on the phone model that you chose earlier.
Activity Verification
You nav e completed this task when you attain these results:
• The new profile is configured in Dev ice >Device Settings >Device Profile as described in
the activity procedure.
• Director) number 2405 is associated with the new device profile andy_dp as described in
the activity procedure.
Lab Guide
© 2010 Cisco Systems, Inc
• H.323 gateway
• PSTN with PS'I N phone
Note The Cisco Extension Mobility service can be activated on multiple servers
Step 4 In Cisco Unified Communications Manager Administration, navigate to Systc
Service Parameters.
Note
It is acommon configuration error to subscribe to the Cisco Extension Mobility service only
at the IP phone and not also at the device profile. In such a situation, you cannot log out of
the phone anymore once you have logged in. By setting the maximum login time to a
relatively low value, you have a back door for this case, because an auto logout is
performed after expiration of the maximum login time. This is a common setup for a lab
environment.
Activity Verification
You have completed this task when vou attain these results:
• The Cisco Extension Mobility service parameters in System >Service Parameters tire
configured as described in the lask.
Note Further venfication will bedone in a later task of this lab exercise.
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Implementing Cisco Unrfied Communications Manager, Part 2(CIPT2) ve.O ©2010 Cisco Systems, Inc
Lab 4-2: Implementing Cisco Extension Mobility
Complete this lab activity to practice what you learned in the related module.
Activity Objective
In this activity. you will implement Cisco Extension Mobility for roaming users. After
completing this activity, you will be able to meet these objectives:
• Configure the sen ice parameters for Cisco Fxtension Mobility
• Create a device profile for a user
• Add an end user
• Associate the end user with the user deviceprofile
• Subscribe IP phones and device profiles to the IP phone service for Cisco Extension
Mobility
Visual Objective
The figure illustrates what vou will accomplish in this activity.
=hpne'-< PI>one2
J Allow roaming
users to log in to
HCP ^T^DHCP any phone and
have personal
settings applied
Required Resources
These are the resources and equipment that arc required to complete this activity:
• One Cisco Unified Communications Manager cluster
• Student PC
• Cisco IP Phones
• Cisco IOS MGCP gateway
Lab Guide 79
© 2010 Cisco Systems. Inc
The local route group ofaroaming phone should be updated. Perform the following steps
for verification:
Step 3 At the phone configuration window, click the View Current Device Mobility
Settings link next to the Device Mobility Mode parameter. Awindow will pop up,
show ing ihecurrent configuration of the phone.
Note
The local route group is nol shown in the pop-up window. You cannot verify that the local
route group has been updated by using the View Current Device Mobility Settings link.
Step 4 Log in to the HO-.v router and enable ISDN debugging with the debug isdn q931
command. Make sure to turn onmonitoring with the terminal monitor command.
Step 5 Shut down the ISDN interface at the branch router so that the gatcwav cannot be
used for TEHO calls to the BR PSTN.
Step 6 From the Phonc3-.v (which is currently roaming to the HQ site) place acall to the
BR PSTN by using the home dial rules (for example, 9 5554444). 'fhe call uses the
IIQ-.v gateway. This indicates that the local route group ofthe phone was updated by
the one configured in the roaming device pool.
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Step 17 Click Save.
Step 18 Click Add New and enter the following parameters in the Device Mobility Into
Configuration window:
• Name: BR_dmi
• Subnet: KU.4.0
• Subnet Mask: 24
Step 19 Mov ethe Branch device pool from the Available Device Pools list to the Selected
Dev ice Pools list.
Note When it is in the home location, Phone2-x uses G.722 for calls to Phone1-x.
Lab Guide
i 2010 Cisco Systems. Inc
Task 1: Configure Device Mobility
In this task, vou will configure the Device Mobility feature so that roaming users can use their
home dial rules, but can aiso use the local route group as abackup path for TEHO PSTN calls.
Activity Procedure
Complete these steps:
Configure Physical Locations
Step 1 Nav igate to System > Physical Location and click the Add New button.
Step 2 In the Physical location Configuration window, enter the following parameters:
• Name: HQpl
• Description: Headquarters
Step 3 Click Save.
Step 4 Click Add New and configure another physical location for the branch office, with
the following parameters:
• Name: BR pi
• Description: Branch
Step 5 Click Save.
Note No DMGs are required. When no DMG is set at the roaming device pool and at the home
device pool, the device-mobility-related settings are not updated. In this lab, globalized call
routing is used. Therefore, there is no need to change the device CSS, AAR group, or AAR
CSS,as theyare the same at allphones.
• Subnet Mask: 24
Step 16 Move the Default device pool from the Available Device Pools list lo ihe Selected
Device Pools list.
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Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v*0©2010 Cisco Systems. Inc.
Lab 4-1: Implementing Device Mobility
Complete this lab activity to practice what you learned in the related module.
Activity Objective
In this activitv. vou will enable the Device Mobility feature to help mobile users who roam
awav from their home location. After completing this activity, you will be able to meet these
objectives:
• Configure Device Mobility
Visual Objective
The figure illustrates what vou will accomplish in this activity.
Required Resources
These are the resources and equipment that arc required to complete this activitv
• Cisco Unified Communications Manager
• Student PC
• Cisco IP Phones
• H.323 gateway
• PSTN with PSTN phone
Lab Guide 75
© 2010 Cisco Systems, Inc
Note The SIP trunk does not need to have access to the RSVP Agent media resource. Therefore,
MRGs and MRGLs do not have to be modified.
Step 4 Nav igate to Device >Device Settings >SIP Profile and click the Add button. mm
Step 5 In the Name field, enter SIPPrecondition, and scroll down lo the Trunk Specific
Configuration section. ^
Step 6 From the Reroute Incoming Request toNew Trunk Based On drop-down menu,
select Never. ' |L
Step 7 From the RSVP Over SIP drop-down menu, select E2E and at theSIPRel IXX
Options, choose Send PRACK if Ixx Contains SDP. 'mt
Step 8 Go to the configuration page ofthe existing trunk called SIPJI'runk.
Step 9 From the SIP Profile drop down menu, select SIP_Precondition and click Save and **
Apply.
Configure the RSVP Bandwidth at the IP WAN Interface That Connects to the Other Pod "**
Step 10 At the subinterface that interconnects the headquarters and the other pod, configure jjL
the bandwidth thatcan bereserved by RSVP as follows:
interface Serial... s«-
ip rsvp bandwidth 4 0
Note Check that the configuration of the other pod is also completed before continuing.
Step 11 Establish one call between your local headquarters phone and the other headquarter
phone at the other pod and keep the call open. Try to set up asecond call by calling
the second headquarter phone for the other pod. The second call should be rerouted
over the PSTN Use the debug isdn q931 command toverify thai the call is sent
through the PSTN. The show seep connections rsvp command can be used toshow
the currently active connections atthe RSVP agent.
Activity Verification
You have completed this task when you attain these results: in
• You configured SIP Preconditions between the two pods as described in the activity
procedure.
You can place one call between the two pods over the SIP trunk and end-to-end RSVP is ^
used for that call as described inthe activity procedure.
When placing an additional call, the PSTN is used as abackup as described in the activity jk
procedure.
Note Make sure lo turn off all of the debug commands at all of the routers (use no debug alt
command).
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Implementing Cisco Unified Communications Manager, Part 2(CIPT2) V8.0 ©2010 Cisco Systems, Inc.
Step 19 Repeat the previous steps for Phonc2-* and Phone3-x
Modifying the RSVP Bandwidth So That All Calls Fail
Step 20 Reconfigure the RSVP bandwidth on one of the two routers to avalue below 40
kb/s. This will make all calls between branch and headquarters fail.
Step 21 Try calls between the headquarters and branch. The calls should be rerouted through
ihe PSTN.
Cleanup
Step 26 Remov ethe (TNB setting at Phone 1-x by clearing the AAR destination mask at
line 1.
Activity Verification
You have completed this task when vou attain these results:
• When intracluster calls are rejected because there is no available bandwidth, the calls are
rerouted over the PSTN as described in the activity procedure.
Note Make sure to turn off all of the debug commands atall of the routers (use the no debug all
command) ^
Activity Procedure
Complete these steps:
Enable End-to-End RSVP to Be Used for Calls Between the Two Pods Using the SIP Trunk
SIP Preconditions-based CAC should allow one Ci.729 call between the two pods.
Step 1 Cio to the configuration page of the existing location called Trunk.
Step 2 Prom fhe Modifv Settings) to Other Locations pane, select the Iliib_None location,
and from the RSVP Setting drop down menu, choose Mandatory (V ideo Desired).
Step 3 Click Save.
Lab Guide 73
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Task 3: Configure AAR and CFNB to Route Calls over the PSTN
If They Are Not Admitted by the Deployed CAC Methods
In this task, you will configure a backup path for calls that are rejected by the previously
implemented CAC methods. These calls will be rerouted over the PSTN using AAR and
CFNB.
Activity Procedure
Complete these steps:
Enable AAR
In the following steps, you will enable AAR by setting the Cisco CallManager service
parameter Automated Alternate RoutingEnabled to True.
Step 1 Navigate to System > Service Parameters and choosethe Cisco Unified
Communications Manager(IOjc.1.1).
Step 2 From the Service drop-down menu, choose Cisco CallManager.
Step 3 Locate the Clusterwide Parameters(System—CCM Automated Alternate
Routing) pane.
Step 4 Set the Automated Alternate Routing Enable parameters toTrue.
Step 5 Click Save.
Step 15 From the Related Links, choose Configure Device and click <;».
Step 16 At the Phone Configuration window, choose Global ess for the AAR CSS.
Note When acall between the HQ and BR sites is not admitted, AAR will be used to place the call
over the PSTN The AAR call will match the Wtranslation pattern first, and then the TEHO
pattern of the local pod. The first option of the route list that is applied to the TEHO pattern is
the TEHO gateway. This, however, cannot be used, because there Is not enough bandwidth
available between the HQ and BR sites. Therefore, the second option of the route list is
used—the local route group.
72 Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 ©2010 Cisco Systems, Inc
Step 19 Click Save.
Note The RSVP configuration has to be performed onboth sides. Unless both sides are
configured for RSVP, the call will fail.
Step 22 Place a call between aheadquarters phone and a branch phone. The call should fail.
Step 23 Use the debug ip rsvp signaling command on the I\Q-x router to see why the
reservation fails. Ilow much bandwidth do you expect to be reserved? How much is
aetuallv reserved0
Note During the call setup phase, the RSVP agents always attempt to reserve an additional 16
kb/s (for signaling). Therefore, in this case, the RSVP bandwidth atthe interface must allow
40 kb/s for the call togothrough. The extra 16 kb/s that that the RSVP agents attempt to
reserve during call setup areimmediately released once the call issetup end-to-end.
Step 24 Change the RSVP bandwidth at the IP WAN interfaces on the IIQ-jc and on the BR-
.i routers to 40 kb/s. Now the calls should go through. If you want, you can retry
with 39kb/s tomake sure that 40 kb/s isthe absolute minimum for one G.729 call to
be allowed. Make surethat youset it back to 40 kb/safterward.
interface serial...
ip rsvp bandwidth 4 0
Note The RSVP bandwidth command has tobemodified on both sides. Unless both sides permit
enough bandwidth for RSVP, thecall will fail. . ^
Step 25 I stablish one call between a headquarters phone and the branch phone and keep the
call open. Try to set up asecond call by calling the branch phone directory number
from the other phone inihe headquarters. The call should fail.
Activity Verification
You have completed this task when you attain these results:
• You modified Cisco Unified Communications Manager CAC between the IIub_None and
the branch locations to use RSVP asdescribed intheactivity procedure.
• RSVP permits one G.729 call between these two locations. Additional calls fail, because of
a lack ofavailable bandwidth asdescribed inthe activity procedure.
Note Make sure to turn off all of the debug commands at all of the routers (use the no debug all
command) ^
Lab Guide
) 2010 Cisco Systems. Inc
dspfarm profile 2 mtp
codec pass-through
rsvp
Note Use the main interface that is connected to the Frame Relay network (PSTN router).
Step 9 At the subinterface that interconnects the headquarters and the branch router,
configure the bandwidth that is allowed to be reserved by RSVP as follows:'
interface Serial...
ip rsvp bandwidth 24
Step 10 Save the configuration to NVRAM.
Step 11 Repeat the above steps (configure Cisco IOS routers to provide RSVP agent MTP
resources and configure the RSVP bandwidth onthe IPWAN interfaces of the
routers) at your BR-x router. When configuring the media resource, use the name
BR-RSVP instead ofHQ-RSVP in the associate profile command.
Add the RSVP Agents (MTPs) in Cisco Unified Communications Manager
Step 12 In Cisco Unified Communications Manager Administration, navigate to Media
Resources >Media Termination Point and click Add New.
Step 13 Verify that the Media Termination Point Type value isCisco IOS Enhanced
Software Media Termination Point.
Step 14 Enter HQ-RSVP for the Media Termination Point Name value.
Note
The location Hob^None and region HQ are applied to the HQ-RSVP media resource
through the device pool Default.
Step 18 Click Copy and change the name to BR-RSVP, the description to HR-a RSVP
Agent, and device pool to BR.
Note
The location Branch and region BR are applied to the HQ-RSVP media resource through the
device pool Branch.
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Step 12 Repeat the previous steps lo apply the llub_Nonc location to the MOH.Only
device pool.
Step 13 Repeat the previous steps to apply the Branch location to the BR device pool.
Step 14 Nav igate lo Device >Trunk and click the Find button.
Step 15 Choose the SIP_Trunk.
Step 16 From the location drop-down menu, choose Trunk.
Step 17 Click Save and reset the trunk.
Activity Verification
You have completed this task when you attain these results:
• You have configured and applied locations as specified in the activity procedure.
• You cannot place more than one call to the other cluster using the SIP trunk (by dialing
851 v-2001. 85 ly-2002. or 852v-300l). The G.729 codec should be in use lor ibis call.
• You cannot place more than one call to the branch phone (by dialing 3001). The (G.729
codec should be in use for this call.
Note RSVP is configured per pair of locations. The setting applies to both directions. Therefore,
the configuration that you apply to one location automatically updates the other location
accordingly .
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© 2010 Cisco Systems. Inc
H.323 gateway
PSTN with PSTN phone
Job Aids
This job aid is available to help you complete the lab activity.
i»
Location Configuration
Name Allowed Bandwidth Applied To Device Applied To Device
Pool
Trunk 24 kb/s
SIPJTrunk
Activity Procedure
Complete these steps:
Add Locations to Cisco Unified Communications Manager
Create new locations as described in the "Location Configuration" table in the .lob Aids section.
Step 1 Navigate to System > Location and click the Find button.
Step 2
Click the location name Hub_None to enter the Location Configuration window.
Step 3 Make sure that the Hub^None location has unlimited audio bandwidth.
Step 4 Click the Add New button.
Step 5 Configure anew location with the following parameters:
• Name: Branch
Apply the newly created locations to devices, through the device pool or directly, as described
inthe Location Configuration" table in the Job Aids section.
Steps Navigate to System > Device Pool and click the Find button.
Steps
Choose Default to enter the Device Pool Configuration window for device pool
Default. •
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Lab 3-2: Implementing CAC
Complete this lab activity to practice what you learned in the related module.
Activity Objective
In this ael.v itv. vou will implement CAC by configuring locations and deploying RSVP agents
to prevent WAN bandwidth oversubscription. You will implement AAR to route calls over the
PSTN ifthev were not admitted bv locations-based CAC. Then you will mplement SIP
Precondition's in order to implement end-to-end CAC for the SIP trunk to the other pod. After
completing this activity, you will be able lo meet these objectives:
• Configure standard locations
• Configure RSVP-enabled locations
i Configure AAR to route calls over the PSTN ifthey arc not admitted by the deployed CAC
methods
Visual Objective
fhe figure illustrates what vou will accomplish in this activity.
Configure
locations. RSVP
agents. AAR, and
' _^- SIP Preconditions
HQ-.
;ۤ
Required Resources
These arc the resources and equipment that are required to complete this activity:
• Cisco Unified Communications Manager
• Student PC
• Cisco IP Phones
Lab Guide
2010 Cisco Systems. Inc
Enable Multicast MOH from Branch Router Flash at the BR-x SRST Router
Step 51 Configure the branch router as follows: Hi
telephony-service
multicast moh 239.1.1.1 port 16384 route 10.x.4.102 El
Note The moh moh-file-name command that is used to enable unicast MOH in SRST mode -
already configured in an earlier lab exercise.
Note The phone now plays the locally generated multicast MOH stream.
' ' . ___ -•*&•
Activity Verification
You have completed this task when you attain these results: Si
• Branch phones can play MOH created by the local SRST router as described in the activity e
procedure. J ^
Hm%
m*
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Step 42 Press the Settings button atPhone3-.v.
Step 43 Choose option 2 Network Configuration.
Step 44 Press 6or scroll down to get to option 6. IP Address.
Step 45 Write down the IP address ofPhone3-.v: 10. .4.
Step 46 Using aweb browser, browse to the IP address of Phone3-.r.
Note If you did not enable web access to the phone earlier, you need to enable it now, in order to
be ableto browse to the phone.
Step 47 Click the Stream 1link to see details about the current RTP stream.
Step 48
The local address should be 239.1.1.1/16384, which indicates that the phone listens
to the multicast MOH stream. Keep the call in this stale so that you hear MOH.
PreventMulticast MOH from Being Sent overthe IP WAN
Step 49 At router HQ-.t. disable multicast routing toward the branch by entering the
following commands:
interface Serial...
no ip pim sparse-dense-mode
Note Use the interface that connects to the BR-x router (IP WAN).
Note
As soon as you enter the above commands, Phone3-x should not play MOH anymore Nor
will it play TOH, because Cisco Unified Communications Manager is unaware that the phone
nolonger receives theMOH audio stream _____
Step 50 At router BR-.r. disable multicast routing by entering the following commands:
interface Serial...
no ip pim sparse-dense-mode
Note Use the interface that connects to the HQ-x router (IP WAN)
interface FastEthernet...
no ip pim sparse-dense-mode
no ip multicast-routing
Note
Mullicast MOH in SRST does not require multicast routing. It simply streams permanently at
the interface that is configured to be used by SRST. If the stream is requ.red on adifferent
mterface (for example, when using aloopback for SRST) the interface or interfaces can be
specified using the route option of the multicast moh command (as shown in the next step)
Lab Guide 65
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m
At this point, the MOH server shares the same region with all other headquarters devices To
limit calls to these other devices to G.729 but allow G.7I Ibetween the MOH server and the
branch phones, the MOH server needs to be placed into aseparate, dedicated region In
addition, the multicast MOH stream that is generated by the MOH server has to be blocked
from the IP WAN. Then multicast MOH can be enabled at the BR-* SRST router.
Allow G.711 Between the MOH Server and Branch Phones
it
Step 29 In Cisco Unified Communications Manager Administration, navigate to System >
Region and click Add New.
k
Step 30 Enter MOH for the name and click Save.
Step 31 Using the Modify Relationship to other Regions pane, allow the G.722 and G711 &
audio codecs to be used to the region HQ by highlighting the region HQ in the
Regions list and choosing G.722/G.711 from the Audio Codec drop-down menu £.
Click Save. mt
Step 32 Using the same technique, also allow the G.722 and G.711 audio codecs for calls m\
between the region MOH and the region BR and for calls within the region MOH.
Allow G.729 only for calls between the regions MOH and Trunks. m
Note
You must click Save after each change in the Modify Relationship to Other Regions pane
The changes will then appear in the Region Relationships pane.
Note By limiting the audio codec to G.729 for calls between regions Trunks and MOH you
effectively disable MOH for these calls. The reason is that the MOH server is only streaming
G.711 multicast MOH, and a multicast stream cannot be transcoded (which would be
required toward the region Trunks). This is desired, because G.729 MOH has only poor
quality, and G.711 must not be sent over the IP WAN (used by the trunks).
Step 36 Navigate to Media Resources >Music On Hold Server and click lind.
Step 37 Choose HQ-SVV-MOH.
Step 38 Change the device pool from Default to MOH_Only.
Step 39 Click Save.
Step 26 Click the Stream I link to see details about the current RIP stream.
Step 27 The Local Address should be 239.1.1.1 /16384. which indicates that the phone listens
lo the multicast MOH stream.
Step 28 Place acall from Phone I-x to Phone3-.v. and at Phone I-x put the call on hold.
Phonc3-.v will play lone on hold only.
Note The reason that Phone3-x will play tone on hold instead of MOH is that the MOH server is
configured for G.711 MOH only (this is the default configuration). However, before changing
to multicast MOH, Phone3-x played MOH. This was possible by using the transcoder media
resource The MOH server is configured with the device pool Default, which applies region
HQ and MRGL HQ_mrgl. This MRGL allows the MOH server to access the transcoder
Such a configuration is not recommended, because if MOH with the G.729 codec should be
permitted, it can be directly enabled on the MOH server (by using the Supported MOH
Codecs service parameter of the Cisco IP Voice Media Streaming Application service)
Using G.729 for MOH, however, is not recommended, because the G.729 codec audio
quality for music is poor; G.729 is designed and optimized for human speech, and does not
work well with music.
Multicast audio streams cannot be transcoded, sobranch phones do not hear MOH
anymore, because the MOH server was configured to use multicast MOH instead of unicast
MOH You will solve this problem by implementing multicast MOH from branch router flash.
interface FastEthernet...
ip pim sparse-dense-mode
Note Use the interface that connects to the voice server network (CUCM-x).
interface FastEthernet...
ip pim sparse-dense-mode
Note Use the interface that connects to the BR-x router (IP WAN).
Step 19 At router BR-*, enable multicast routing using the following commands;
ip multicast-routing
interface Serial...
ip pim sparse-dense-mode
Note Use the interface that connects to the HQ-x router (IP WAN).
interface FastEthernet...
ip pim sparse-dense-mode
Note
Multicast routing is now enabled for the voice server network, the headquarters phone
network, the branch phone network, and the link between HQ-x and BR-x (IP WAN).
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Stepl Initiate an ad hoc conference from Phone3-jr (adding Phonel-* and Phone2,v>.
Step 2 At each IP phone, press the ?button twice. Phonel-.v and Phonc2-.v should use the
G729 codec and Phone3-j should use the G.711 codec. This is because the
conference bridge is at the branch and G.7I I is only allowed locally at the branch
butnotbetween branch and headquarters. Keep the call open.
Step 3 At BR-.r. enter the show dspfarm cisp all command. You should see three used
connections representing the three conference participants.
Step 4 r.nd the conference. Verifv thai all DSP resources arc freed by entering the sho»
dspfarm dsp all command again.
Step 5 Repeat the previous steps but initiate the conference from Phone I-x or Phonc2-.v.
1his time Phonel-.v and Phone2-Jc should use G.7I I and Phone3,v should use G.729.
The show dspfarm dsp all command will indicate that no conference resources are
used at BR-.v. Issue the same command at I\Q-x and you will see atranscoder
session for the connection of Phone.V.v to the conference bridge.
Activity Procedure
Complete these steps:
Enable Multicast MOH in Cisco Unified Communications Manager
Step 1 In Cisco Unified Communications Manager Administration, navigate to Media
Resources >Music On Hold Audio Source and click the Find button.
Step 2 Choose the onlv available audio source and verify that the Play Continuously check
box is checked.
Lab Guide
© 2010 Cisco Systems, Inc.
Step 17 Enter HQ_mrg for the name of the MRG.
Step 18 Enter HQ SW Conference Bridge for the description.
Step 19 From the Available Media Resources pane, add HQ-SW-CFB to the Selected Media
Resources list.
In these steps, you will configure dilTerenl MRGLs that allow IP phones to use their local t*
conference bridge.
Step 22 Navigate to Media Resources> Media Resource Group List. **
Step 23 Click the Add Newbutton. __
Step 24 Enter HQ_mrgl for the name of the MRGL.
Step 25 From the Available Media Resource Groups pane, add the HQ-SW-CFB and m,
General_mrg MRGs to the Selected Media Resource Groups list.
Step 26 Click Save. »
Step 27 Repeat the previous steps to add the other MRGL, as described in the "Media
Resource Group List Configuration1" table in the Job Aids section. H
Assign MRGLs to Devices
In these steps, you will assign the newly created MRGLs to devices (phones, trunks and "*
gateways) by configuring the appropriate MRGL in the available device pools. ' m,
Step 28 Navigate to System > Device Pooland click Find.
Step 29 Choosethe device pool Default. tt
Step 30 From the Media Resource Group List drop-down menu, choose IIQ_mrgl.
Step 31 Click Save. W
Step 32 Reset the device pool. m
Step 33 Repeat the previous steps for the other two device pools, assigning the MRGLs as
described in the "Device Pool Configuration" table in the Job Aids section. M
Activity Verification
You have completed this task when you attain these results: •
• BR-.v provides aconference hardware media resource (BR-HW-CFB) lhat is registered m
with Cisco Unified Communications Manager. Perform the following steps at router BR-v ""
to verity the hardware media resource configuration and status-
mh
Step 1 Enter the show seep command. Verify that the Conferencing Oper Stale is ACTIVE
and that the TCP Link Status is CONNECTED. m
Step 2 Enter the show dspfarm profile 1command. Verify the stains, number ofavailable
resources, and the listof supported codecs. m
• Conferences that are initiated by headquarters phones use the software eonlerenee bridge
that ,s located at the headquarters; branch phones use the hardware conference bridge that J£
is located at the branch. Verify this by performing the following steps- "
60 Imptemenflng Cisco Unified Communications Manger, Part 2(CIPT2) v8.0 ©20)0 Cisco Systems, Inc.
Step 3 To configure the router DSP resources that are to be used as ahardware conference
bridge, enter this sequence ofcommands:
voice-card 0
dspfarm
dsp services dspfarm
Create MRGs
Step 15 Nav igate to Media Resources >Media Resource Group.
Step 16 Click the Add New button.
Lab Guide 59
© 2010 Cisco Systems. Inc
Step 6 From the Transcoder Type drop-down menu, choose Cisco IOS Knhanced Media
Termination Point.
Step 9 Apply the device pool Default to the transcoder. This device pool is configured with ^
the region HQ.
Step 10 Click Save. j*
Step 11 Reset the newly created transcoder.
Step 12 Verify the registration status. Itshould say Registered with Cisco Unified ""
Communications Manager 10jr. I. I.
Activity Verification **
You have completed this task when you attain these results: §^
• HQ-* provides atranscoding hardware media resource (HQ-HW-XCD) that is registered
with Cisco Unified Communications Manager. Perform the following steps at router HQ-* H
toverify the hardware media resource configuration and status:
Step 1 Enter the show seep command. Verify that the Transcoding Oper State is ACTIVE Bl
and that the TCP Link Status is CONNECTED.
Step 2 Enter the show dspfarm profile I command. Verify the status, number of available &
resources, and the list of supported codecs.
• Branch phones can now join conferences on the G.711-only software conference bridge 1*
even though they are not allowed to use the G.711 codec over the IP WAN. Verify this by
performing the following steps: m
Step 1 Set up an ad hoc conference with Phonel-*, Phone2-*, and Phone3-.v as members.
Step 2 At each IP phone press the ?button twice. Phonel-* and Phone2-.r should show the ^
G.711 codec being used for the call, while Phone3-* shows G.729.
Step 3 At HQ-.v. enter the show dspfarm dsp all command. You should see two used *
connections representing the two call legs ofthe transcoder (G.711 tothe software
conference bndge and G.729 to Phone3-*). §fc
Task 4: Implement a Hardware Conference Bridge m
In this task, you will configure alocal hardware conference bridge at the branch You will
implement MRGs and MRGLs to ensure that IP phones use the local conference media Mi.
resource. am
Configure aCisco IOS Router as aHardware Conference Media Resource for the Branch
Step 1 Connect to your BR-*. f&
Step 2 Enter configuration mode.
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Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
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Task 3: Implement Transcoders
In this task vou will implement atranscoder at the headquarters in order lo allow branch users
to join confe'renccs on aG.711-only software conference bridge, even though branch phones
are not allowed to use G.711 over the IP WAN. The transcoder will transcode the (..729 audio
stream that is received from branch phones to aG.711 stream toward the software conference
bridge and vice versa.
Activity Procedure
Complete these steps:
Configure a Transcoder Media Resource in Cisco IOS Software
Configure the IIQ-.v router as a transcoder resource.
Step 1 Connect to \our HQ-.* and enter configuration mode.
Step 2 To configure router DSP resources to be used as atranscoder. enter this sequence of
commands:
voice-card 0
dspfarm
dsp services dspfarm
i
Note The highest possible SCCP version that can be specified in the seep ccm command
depends on the Cisco IOS Software release that is used on the router.
seep
Lab Guide
)2010 Cisco Systems. Inc
Step 35 Change thedevice pool from Default to BR.
Step 36 Click Save.
Step 37 Reset the gateway in Cisco Unified Communications Manager and reset the MGCP
process at the BR gateway by entering the no mgcp command, followed by the
mgcp
morn command.
rnmmnniH
I*
Note
You already verified the device pool configuration (and hence the region assignment) of the
software media resources in the previous task. All software media resources are configured
with the device pool Default
Activity Verification
You have completed this task when you attain these results:
• Place test calls between the following phones and while on acall press the 7button on the
IP phone two times. The IP phone will display call information that includes the codec thai
is used for the call:
— Phone 1,r or Phone2-* and the PSTN (for example, 0 112): This call should use
G.711.
— Phone I-x or Phone2-j; and any phone that is located in the other pod (for example
dial851y2001):ThiscallshoulduseC..729. '
— Phonel-j and Phone2-.r: This call should use G.722.
— Phonel-.t orPhone2-;r and Phone3-*: This call should use G.729.
— Phone3-.v and the PSTN (for example, dial 9 911): This call should use G.711.
— Phone3-.v and any phone that islocated in the other pod (for cxumnle dial 851 v
2001 ):This call should use G.729.
Tip You can also view information about active calls of an IP phone by using aweb browser to
browse to the IP address of the IP phone. The built-in web server of the phone provides
information about active RTP streams.
The built-in web server is disabled by default. You need to enable it when you want to
examine the information that is provided by the built-in web server. The built-in web server
can be enabled at the phone configuration page: setthe Web Access parameter to
Enabled
Note
You cannot add Phone3-x to aconference anymore. The only available conference bridge
(HQ-SW-CFB) is asoftware conference bridge running on Cisco Unified Communications
Manager. This software conference bridge supports G.711 only, Because Phone3-x is in
region BR and Ihis region is not permitted to use G.711 to region HQ (where the software
conference media resource is in), Phone3-x cannot join conferences anymore
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Step 17 Verify that the region issetto HQ.
Step 18 At the relaled links, choose Back To Find/List and click Go.
Step 19 Choose the Branch device pool.
Step 20 Change the region lo BR.
Step 21 Click Save.
Step 22 Reset the device pool.
Step 23 Click the Add New button and configure anew device pool with following
parameters:
• Name: Trunks
• Cisco Unified Communications Manager Group: Default
• Local Route Group: HQrg
Note The local route group at the trunk is required in order to allow received TEHO calls that are
to besent to the BR gateway to bererouted via the backup path (standard local route
group). Until now, the trunk had the device pool Detault applied, which also has the local
routegroup HQ_rg configured. ^^_
• Date-TimeGroup: CM Local
• Region: Trunks
• SRST Reference: Disable
Note Regions cannot be directly applied to devices. You have to create different device pools with
regions and then apply the appropriate device pools to the devices.
Step 7 Enter Trunks for the name ofthe new region and click Save.
Step 8 Using the Modify Relationship to other Regions pane, allow the G.729 audio codec
for calls within region Trunks by highlighting Trunks in the Regions list and
TJ-.»" /Villi- llJltliin Hr^x^-^^.^- T 1.1 I • II- • .• »», -
choosing G.729 from the Audio Codec drop-down menu. Click Save.
•to
Step 9 Using the same technique, allow the G.729 audio codec for calls between region
Trunks and region HQ.
Note
You must click Save after each change in the Modify Relationship to Other Regions pane
The changes will then appear in the Region Relationships pane.
Step 11 Enter BR for the name ofthe new region and click Save.
Step 12 Allow G.722/G.711 for calls within region BR.
Step 13 Allow G.729 for calls between regions BR and Trunks.
Step 14 Allow G.729 for calls between regions BR and HQ.
Create and Configure Device Pools
In these steps, you add anew device pool for the trunks and update the existing device pools
with the new regions, as described in the "Device Pool Configuration" table in .he Job Aids
otciions.
Step 15 Navigate to System >Device Pool and click the Find button.
Step 16 Choose the Default device pool.
54 Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 >2010 Cisco Systems. Inc.
Task 1: Enable Software Media Resources on Cisco Unified
Communications Manager
In this task vou will enable the Cisco IP Voice Media Streaming Application service, which
provides several software media resources running on Cisco Unified Communications
Manager. You will change the default names and descriptions of these media resources.
Activity Procedure
Complete these steps:
Verify That the Cisco IP Voice Media Streaming Application Service Is Activated
Step 1 Log in to Cisco Unified Serviceability and navigate to Tools >Control Center-
Feature Services.
Step 2 Verify that the Cisco IP Voice Media Streaming App service is activated and
running at CUCMl-.v.
Note This service provides the following software media resources: Annunciator, Conference
Bridge, Media Termination Point, and Music on Hold Server.
Name
Description
Media Resource
Activity Verification
You have completed this task when you attain these results:
• The Cisco IP Voice Media Streaming Application service is activated.
. The following software media resources are registered with CUCMl-.v: Annunciator.
Conference Bridge. Media Termination Point, and MOH Server.
Lab Guide
J2010 Cisco Systems. Inc.
• H.323 gateway
• PSTN with PSTN phone
Job Aids
These job aids are available to help you complete the lab activity.
Region Configuration
HQ Trunks BR
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Lab 3-1: Implementing Bandwidth Management
Complete this lab activity to practice what you learned in the related module.
Activity Objective
In this activitv. vou will configure multicast MOH from branch router flash, regions, local
conference bridges, and Iranseoders to reduce bandwidth requirements on the IP WAN. After
completing this activ ity. you will be able to meet these objectives:
• fnable software media resources on Cisco Unified Communications Manager
• Configure regions
• Implement transcoders
• Implement a hardware conference bridge
• Implement multicast MOH from branch router flash
Visual Objective
The figure illustrates what vou will accomplish in this activity.
Implement
conference bridge
and transcoder
Use low-bandwidth
codecs only in WAN
Use low-bandwidth
codecs only in WAN.
Implementlocalconference bndge;
npiement multicast MOH from branch
router flash
Required Resources
These arc the resources and equipment that are required to complete this activity:
• Cisco Unified Communications Manager
• Student PC
• Cisco IP Phones
Lab Guide
© 2010 Cisco Systems. Inc
Step 4 When prompted for the IP address of the FTP server, enter the IP address you wrote
down in Step 2.
Step 5 For the source filename, enter moh.au.
Steps Confirm the destination filename (moh.au) and wait for the file to be copied.
Step 7 Verify that the moh.au file isstored in flash by entering the show Hash command.
Enable MOH in SRST Mode
Step 8 Enable MOH at the branch by entering the following commands at BR-.v (in
configuration mode):
telephony-service
moh moh.au
Step 1 Place a call between aheadquarters phone and the branch phone.
Step 2 At the branch phone (Phone3-^). putthecall on hold.
Step 3 The headquarters phone should play MOH coming from the branch router.
Note
When you are finished, make sure to remove the access-list that you entered in an earlier
task to break the connection between BR-x and CUCMI-x from the serial interface atrouter
HQ-x. Verify that the Phone3-x re-registers with Cisco Unified Communications Manager.
&
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Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 >2010 Cisco Systems, Inc.
Activity Verification
You have completed this task when you attain these results:
• To verify the SRST Fallback configuration, enter the show telephony-service command on
the branch router BR-.v.
• To verity that the current flies are accessible to IP phones, enter the show telephony-
servicetftp-bindings command.
• To verify that SRST isworking, follow these steps:
Step 1 Break connective lo Cisco Unified Communications Manager by reapplying access
list 100 in the incoming direction atthe interface ofthe HQ-.r router that connects to
the BR-.v router.
Step 2 Place acall from aheadquarters phone to the branch phone (300I). The call should
work: the calling party number should be the 10-digit PSTN number ofthe
headquarters phone.
Step 3 Place acall from Phone3-.r to aheadquarters phone (use internal dialing: 2001 or
2002). The call should work: the calling party number should be the I0-digit PS IN
number of the branch phone.
• lo verifv that ihe learned configuration was saved, display the configuration ofthe router:
— Enter the show running-config command and verify that you see an cphone-dn and
cphonc in the configuration.
Note The next time the phone registers with Cisco Unified Communications Manager Express.
Cisco Unified Communications Manager Express uses the stored configuration instead of
learning the phone configuration using SNAP. In order to configure a phone with features
that cannot be learned by SRST. you can preconrigure the ephone-dn only (and then the
ephone is learned) or ephone-dn and ephone _
Activity Procedure
Complete these steps:
Activity Procedure
Complete thesesteps:
Remove SRST Configuration from the Branch Router
Step 1 Log in tothe BR-j router and enter configuration mode.
Step 2 Delete the SRST command by entering the following command:
no call-manager-fallback
Note The dial peers and translation profiles that are configured in the standard SRST lab will be
reused for Cisco Unified Communications Manager Express in SRST Fallback mode.
Configure Cisco Unified Communications Manager Express in SRST Mode on the Branch
Router
In these steps, you will configure Cisco Unified Communications Manager Lxpress in SRST
mode for the branch router.
Step 3 Enable Cisco Unified Communications Manager Express in SRSI mode by entering
the following commands:
telephony-service
ip source-address 10.x.250.102 port 2000
system message CUCME in SRST Mode
max-ephones 5
max-dn 5
Note The keyword all at the end of the srst mode auto-provision command causes the router to
save the learned ephone and ephone-dn configuration.
Note The create cnf-files command makes the router generate configuration files that,
required by SCCP phones.
end
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Implementing Cisco Unrfied Communications Manager. Part 2(CIPT2>v8.0©2010 Cisco Systems, Im,
Lab 2-2: Implementing Cisco Unified
Communications Manager Express in SRST
Mode
Complete this lab activity to practice what you learned in the related module.
Activity Objective
In this activitv vou will configure Cisco Unified Communications Manager Express in SRST
mode to provide basic telephony services lo phones that lost the connection to Cisco Unified
Communications Manager. In addition, you will enable MOII. After completing this activity,
you will be able to meet these objectives;
• Configure Cisco Unified Communications Manager Express in SRST fallback mode
• Configure MOI 1on Cisco Unified Communications Manager Express
Visual Objective
The figure illustrates what you will accomplish in this activity.
Required Resources
These are the resources and equipment that arc required to complete this activity:
• Cisco Unified Communications Manager
• Student PC
• Cisco IP Phones
Lab Guide
& 2010 Cisco Systems, Inc.
Step 16 Place atest call to the branch phones of the other pod using site-code dialing (852v
300I).
• You can place outgoing calls to the PSTN from Phone3-x The calling party number should
always be shown as 10-digit PSTN number at the PSTN phone. Make sure to place test
callsto the following types of destinations:
— Local destinations, for example by dialing 9-555-5678
— National destinations, for example by dialing 9-1-606-555-1234
— International destinations, for example by dialing 9-011-44-555-666-7777
— Emergency (911 and 9-911)
• You can call Phone3-.r from headquarters phones by using the internal directory number of
Phone3-.T(300I).
Note CFUR, which is required at the main site in this scenario, was already configured in the
previous task. In this task you enabled incoming PSTN calls sothat received CFUR calls
can be routed to the internal number of Phone3-x.
From Phone3,v. you can use the internal numbering plan to reach sites within the pod .
the other pod asdescribed in the activity procedure.
Caution When you are finished, make sure to remove the access list at HQ-x that you entered in an
earlier task to break the connection between BR-x and CUCMI-x from the serial interface at
router HQ-x. Verify that the Phone3-x re-registers with Cisco Unified Communications
Manager.
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Si
Configure Digit Manipulation at the Branch SRST Router for Outgoing Calls
In these steps, you will configure digit manipulation to ensure that PSTN format is used for the
calling partv numbers onoutgoing calls.
Step 7 Configure atranslation profile, which adds 52r555 to the four-digit directory
numbers starting with 3by entering the following commands:
voice translation-rule 2
rule 1 /'3/ /52X5553/
exit
voice translation-profile pstn-out
translate calling 2
exit
Step 8 Bind the translation rule to the voice port that connects to the PSTN by entering the
following commands:
voice-port 0/0/0:23
translation-profile outgoing pstn-out
exit
Lab Guide
© 2010 Cisco Systems. Inc
at
Configure the Branch SRST Router toAllow Outgoing Calls
In these steps, you will add adestination pattern to the existing POTS dial peer Iin order to S*
allowoutgoing PSTN calls.
Step 5 At BR-.t. enter the following commands in configuration mode: •
dial-peer voice 2 pots
destination-pattern 9011T Ifc
prefix Oil
port 0/0/0:23 Jfe
Note
The ISDN switch type that is used at BR-x is primary-ni. This switch type automatically sets
the number type to international when the called number starts with 011 and has 12 more
digits, which can be the case in this lab. The PSTN, however, does not allow the type of
number to be used atthe BR site; only prefixes should be used. The shown isdn map
address command instructs the BR-x gateway not to automatically set the type of number to
international.
Step 6 Verify that outgoing calls are working by calling 9011 55 Six 5552001 from
Phone3-.r. Also try placing acall to the PSTN phone by dialing any valid PSTN
number (for example, 91606 555 4444). Note that the calling party number
displayed on the PSTN phone is the four-digit internal directory number olThoncl-v
Note At this stage, branch phones are able to place calls to the PSTN. This includes calls to
headquarters phones if the headquarters phones are dialed by their PSTN numbers The
caU'ng party numbers of outbound PSTN calls use internal directory number format
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Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 ©2D10 cisco Sys(emSi ,nc
Step 2 Verifv using the debug isdn q931 command whether the call hits the HR-* gateway.
The call arrives at BR-.v. but DID is not enabled and the called party number is a 10-
digit number and not a4-digit director) number.
Note Gateway BR-x accepts the call and, because DID is not enabled, it waits for dialed digits
(two-stage dialing) If you manually enter 3001 at this stage, Phone3-x will ring
Activity Procedure
Complete these steps:
Configure the Branch SRST Router toAllow Incoming Calls
In these steps. >ou will configure an inbound dial peer to allow incoming calls to be routed
correcllv.
Step 1 l.nable DID for the voice port that connects to the PSTN by entering the following
commands:
dial-peer voice 1 pots
incoming called-number 52x5553...
direct-inward-dial
port 0/0/0:23
Step 2 Configure atranslation profile to manipulate the incoming called number from the
PSTN (the complete PSTN number 52.x 55530(11) to the four-digit directorv number,
bv entering the following commands:
voice translation-rule 1
rule 1 /*52x5553/ /3/
exit
voice translation-profile pstn-in
translate called l
Step 3 Bind the newly created translation profile to the voice port that connects lo the
PSTN bv entering the following commands:
voice-port 0/0/0:23
translation-profile incoming pstn-in
Note
At this stage, branch phones are reachable by their PSTN numbers. CFUR from the
headquarters should now work
Step 4 Verifv that incoming calls are working by calling 3001 from PhoneKt or Ph0nc2,v.
Note 'that the calling parly number that is displayed at Phone3,v is the PS IN number
of the calling phone.
Lab Guide
© 2010 Cisco Systems. Inc
Task 3: Implement a Dial Plan in Cisco Unified
Communications Manager Supporting Outbound Calls Durina
SRST Mode y
In this task, you will configure CFUR for remote phones to allow phones that are located at the
main siteto call remote phones viathe PS'fN in SRST mode.
Activity Procedure
Complete these steps:
Adjust CFUR Service Parameters
In these steps, you will adjust the CFUR-Max-Hop-Counter service parameter.
Step 1 Navigate to System >Service Parameters and choose the Cisco Unilied
Communications Manager publisher (IOjc.1.I).
Step 2 From the Service drop-down menu, choose the Cisco CalfManager service.
Step 3 In the Clusterwide Parameters (Feature—Forward) pane, change Ihe Max Forward
UnRegistered Hops toDN parameter to2 (default is0).
Step 4 Click Save.
• Destination: +6651x5553001
• CSS: GlobaI_css
Step 8 Click Save.
Note SRST mode will be active when IP connectivity between HQ and BR sites is broken. Calls
from HQ to BR will use the configured CFUR settings (destination and CSS). The CFUR
destination will match the VH translation pattern first, and then the \+6652x[2-9]XXXXXX
TEHO pattern. The first option of the route list that is applied to the TEHO pattern is the BR
gateway This, however, cannot beused, because IP connectivity between HQ and BR is
broken {the MGCP gateway isdown) Therefore, the second option of the route list is
used—the local route group.
Activity Verification
You have completed this task when you attain these results:
• To verify that Cisco Unified Communications Manager routes calls to the unregistered
numbers of phones that are in SRST mode, perform the following steps:
Step 1 Place acall from one ol'your headquarters phones to 300I. The call will fail.
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Implementing Cisco Unified Communications Manager, Part 2<CIPT2)v8.0 ©2010 Cisco Systems, Inc.
Step 2 Enter the following commands to enable and configure the SRST feature:
call-manager-fallback
ip source-address 10.x.250.102
max-dn 1 dual-line
raax-ephones 1
Activity Verification
You have completed this task when you attain these results:
• To verifv that SRSI' isworking on your branch router, perform the following steps:
Step 1 i; nter the debug ephone register command to start debugging.
Step 2 Fnter the terminal monitor command.
Step 3 To break connectivity to Cisco Unified Communications Manager, enter the
following commands at HQ-.v in global configuration mode:
access-list 100 deny ip any host 10.x.1.1
access-list 100 permit ip any any
Note Use the interface that connects the HQ-x route with the BR-x router
Step 4
Your Cisco IP phone in the branch should register with the BRI-jc SRST router. This
is indicated by the text "CM Fallback Service Operating" at the bottom of the phone
display.
Step 5 At the BR-.t gateway, you will see debug output indicating that the phone registered
with the SRST gateway. The last message should be "ephone-
l|l]:SkinnyCompleteRegistration."
Step 6 When you are finished, turn off all ofthe debug commands at all of the routers
using the nodebug all command.
Lab Guide
© 2010 Cisco Systems, Inc.
• HJ23 gateway
• PSTN with PSTN phone
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Implementing Cisco Unified Communications Manager. Part 2(CIPT2) v8.0 ©2010 Cisco Systems, Inc.
Lab 2-1: Implementing SRST and MGCP Fallback
Complete this lab activity to practice what you learned in the related module.
Activity Objective
In this activ itv vou will configure Cisco Linified SRST to provide call survivability for Cisco
IP phones, and MGCP fallback for gateway survivability. After completing this activity, you
will be able lo meet these objectives:
• Configure SRSI gateways in Cisco Unified Communications Manager
• Configure aCisco IOS gateway for MGCP fallback and SRST
. Implement adial plan in Cisco Unified Communications Manager supporting outbound
calls during SRST mode
• Implement adial plan at the SRST gateway supporting inbound and outbound calls when in
MGCP fallback or in SRST modeor both
Visual Objective
The figure illustrates what you will accomplish in this activity.
Required Resources
These are the resources and equipment that are required to complete this activity:
• Cisco Unified Communications Manager
• Student PC
• Cisco IP Phones
Lab Guide 39
>2010 Cisco Systems, Inc
Configure Calling Party Transformations for TEHO Calls
Step 2 Create calling party transformation patterns for the HQ gateway. Refer to the
"Localization ofCalling Party During Call Egress for Outbound TE110 Calls" table
of the Task Job Aids.
Note The gateway isalready configured with a calling party transformation CSS that has access
to the partition that you applied to the newly created transformation patterns.
Step 3 Place test calls to TEHO PSTN destinations located at the other pod. Use the debug
isdn q93I command at the gateways ofthe other pod to verify that the TEHO
gateway is used. Make sure that for TEHO calls through the HQ gateway, the calling
number isthe number ofthe actual caller (in international format ifthe call comes
from the BR site ofthe other pod and in national format ifthe call comes from the
HQ site ofthe other pod) while the calling number for TEHO calls through Ihe BR
gateway isalways the number ofthe BR attendant (52^5553001).
Verify PSTN Backup for TEHO to the Other Pod
Step 4 Shut down the serial interface that connects your pod with the other pod.
Step 5 Repeat placing TEHO test calls from the HQ site and from the BR site to Ihe PSTN
destination located at the other pod. Although IP connectivity is broken, the calls
should still work, because they are rerouted via the second option ofthe route list:
the local route group. Use the debug isdn q931 command to verify that the call is
set up using the local PSTN gateway.
Step 6 Usethe no shutdown command on the serial interface.
Activity Verification
You have completed this taskwhen you attain this result:
• You can place TEHO calls to the other pod as described in the activity procedure.
• When IP connectivity between the two pods is broken, the local gateway is used as a
backup asdescribed intheactivity procedure.
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Implementing Cisco Unified Communications Manager. Part 2(CIPT2) v8.0 ©2010 Cisco Systems. Inc.
Task 6: Implement TEHO Between Pods
In this task, vou will configure TEHO between your pod and the other pod.
Task Job Aids
These job aids arc av ailable to help you complete the lab task.
Route Patterns for Outbound TEHO Calls
Localization ofCalling Party During Call Egress for Outbound TEHO Calls
Calling PartyTransformation Pattern Configuration
Note
The HQ PSTN allows remote calling numbers to be sent. The BR PSTN does not allow
other calling numbers than the number that is assigned to the PSTN line. Therefore, TEHO
calls from the other pod to HQ are configured to use the numbers of the other pod for the
calling number while calls going through the BR gateway have already been configured in
the previous task to use the number of the BR attendant (3001) for the calling number it the
calling number is not in the locally assigned DID range. Further, the HQ PSTN requires
number types to be set, while the BR PSTN expects 10-digit calling numbers without
number types ^ ___
Activity Procedure
Complete these steps:
Lab Guide
2010 Cisco Systems. Inc
Configure TEHO Route Patterns
Step 3 Create route patterns for TEHO calls. Refer lo the "Route Patterns for Outbound
TEHO Calls" table of the Task Job Aids.
Note The gateways arealready configured with a calling party transformation CSSthathas
access tothe partition that you applied tothe newly created transformation patterns.
Step 5 Place test calls toTEHO PSTN destinations. Use the debug isdn q931 command to w
verify that the TEHO gateway is used. Make sure that for IEl 10 calls through the mt
HQ gateway, the calling number is the number ofthe BR phone (in international wWi
format) while the calling number for TEHO calls through the BR gateway isnot the Mb
IIQ number but the number ofthe BR attendant (52x5553001).
Verify PSTN Backup for TEHO Within the Pod 8
Step 6 Shut down the ISDN interface at the IIQ gateway and try placing aTEI10 call from m
the BR phone. Because the primary path (through the TEHO gateway) does not •"
work, the local gateway (BR) should beused asa backup.
Step 7 Use the no shutdown command on the ISDN interface. 8
Step 8 Shut down the ISDN interface at the BR gateway and try placing aTEHO call from *&
the HQ phone. Because the primary path (through the TEHO gateway) does not '*im
work, the local gateway (HQ) should be used as a backup.
Step 9 Use the no shutdown command on the ISDN interface. m
Activity Verification w
You have completed this task when youattain this result:
• You can place TEHO calls within your pod as described in the activity procedure. 8
• When IP connectivity between the two siles is broken, the local gateway is used as a B
backup as described in the activity procedure. %m
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Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 ©2010 Cisco Systems, lire
Task 5: Implement TEHO Within Your Pod
In this task, you will configure TEHO within your pod.
Partition xform-cg_BR-out
Description: BR (from HQ)
Calling PartyTransform Mask: 52x5553001
Note The HQ PSTN allows remote calling numbers to be sent. The BR PSTN does not allow
other calling numbers than the number that is assigned to the PSTN line Therefore, TEHO
calls from BR to HQ are configured to use the BR number as the calling number while calls
going through the BR gateway are configured to use the number of the BR attendant (3001)
for the calling number if the calling number is not of the locally assigned DID range. Further,
the HQ PSTN requires number types to be set while the BR PSTN expects 10 digit calling
numbers without number types ^
Activity Procedure
Complete these steps:
Configure Route Lists with the TEHO Gateways as First Option and the Local PSTN
Gateway as Backup
Step 1 Create aroute list that is called TEHO-HQ_rl and add the HQj-g and the Standard
Local Route Group item to the route list. Make sure that the HQ_rg is listed first.
Step 2 Create aroute list that is called TUIO-BR.rl and add the BR_rg and the Standard
focal Route Group item to the route list. Make sure that the BR_rg is listed first.
Lab Guide 35
© 2010 Cisco Systems. Inc
Step 11 Make sure that the callingnumberis shown with the internally used directory-
number and a site-code dial prefix at the receiving phone (85 ly-2001 and 851v-2002
for calls that are received from HQ phones of the other pod and 852_y-300l for calls
that are received from the BR phone of the other pod).
Note This is the serial interface at the HQ router that is configured with IP address 10.zx.101 and
subnet mask 255.255.0.0.
As mentioned earlier, x is the number of your pod, and y is the number of your partner pod
(in the same group). Groups are pods 1 and 2, pods 3 and 4, pods 5 and 6, and pods 7 and
8. A z in an IP address stands for the pod numbers of your pod and your partner pod. They
are listed in ascending order. Examples: for pod 1, x=1, y=2, and z=12; for pod 2, x=2, y=1,
and z=12; for pod 3, x=3, y=4, and z=34, for pod 4, x=4, y=3, and z= 34, and so on.
Step 13 Continue placing test calls from the HQ site and from the BR site to the other pod
using intersite dialing, Although IP connectivity is broken, the calls should still
work, because they are rerouted via the second option of the route list: the local
route group. Use the debug isdn q931 command to verify that the call is set up
using the local PSTN gateway. Verify that the calling number is still shown with the
internally used directory number and a site-code dial prefix.
Step 14 Use the no shutdown command on the serial interface.
Activity Verification
You have completed this task when you attain this result:
• You can place calls to and receive calls from the other pod using site-code dialing as
described in the activity procedure.
• When IP connectivity between the two pods is broken, the PSTN is used as a backup as
described in the activity procedure.
• The calling number is always shown with the intemally used directory number and a site-
code dial prefix as described in the activity procedure.
i.
Note At this stage, you are ready to send calls to the other pod using globalized called and calling
numbers, in order to process the calls that are received from the other pod, you have to
change the received called number from globalized format to the internally used directory
number
This called number format change could be done with significant digits set to 4, configured
at the SIP trunk. However, as you want to use the SIP trunk also for TEHO in a later lab
task, the called number cannot be reduced to a four-digit number in general, only if the call
was placed to an internal phone and not when the call was placed to a TEHO destination.
Therefore, you will use translation patterns to modify the called number of inbound calls that
are received through the SIP trunk.
Step 5 Create translation patterns for received intersite calls. Refer to the "Changing the
Called Number of Calls Received Through the SIP Trunk" table of the Task Job
Aids,
Configure the SIP Trunk with a CSS That Has Access to Internal Phones
StepS Apply the CSS Ininkjjss to the SIP trunk.
Step 7 Reset the trunk.
Step 8 When the configuration of the other pod has finished, you can start placing test calls
using intersite dialing. Dial 8-51v-2001 and 8-5ly-2002 to reach the HQ phones of
the other pod. dial 8-52v-300l to reach the BR phones of the other pod.
Note When receiving calls from the other pod, you will see the calling number in globalized format
(+55-51y-555-2001 and +55-51y-555-2002 for calls from the HQ phones of the other pod
and +66-52/-555-3001 for calls from the BR phone of the other pod).
In order to indicate that the call is coming from an interconnected site, the configuration will
be changed in the next steps so that the calling number is shown with the internally used
site code.
Configure the Calling Number of Intersite Calls to Be Shown with Site Codes
Step 9 Create calling party transformation patterns for IIQ and BR phones. Refer to the
"Localization of Calling Party DuringCall Egress for Inbound Intersite Calls" table
of the Task Job Aids.
Note The phones are already configured with a calling partytransformation CSS that has access
to the partition that you applied to the newlycreated transformation patterns.
Step 10 Place lest callsbetween thetwo pods using intersite dialing. Make surethatyou
place calls in both directions. Dial 8-5!v-2001 and8-5l.y-2002 to reach the IIQ
phones of the otherpod. dial 8-52v-300l to reach the BRphones of the otherpod.
Localization of Calling Party During Call Egress for Inbound Intersite Calls
Activity Procedure
Complete these steps:
32 Implemenling Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 © 2010 Cisco Systems. Inc.
Task 4: Configure Intersite Calling
In this task, you will configure intersite calling over the SIP trunk using site codes.
These job aids are available to help you complete the lab task.
Globalization of Called and Calling Parties During Call Ingress for Outbound
Intersite Calls
In order to allow callbacks to globalized numbers, you have to add a \+! translation pattern.
The translation pattern will not change the called number, but only modifies the calling
number by applying the external phone number mask. Further, the translation pattern has a
CSS that has access to the \+! route pattern.
By adding such a translation pattern, you make sure that the calling number of the outbound
callback is globalized during call ingress and that you then match the \+! route pattern.
Applying the external phone number mask at the route pattern instead of the translation
pattern does not work before the configured global transformations are based on globalized
numbers. Digit manipulation that is configured at the route pattern and at the route list is
ignored by global transformations. Global transformations are based on the pre-transformed
number (that is, the number as it looks when hitting the route pattern), not on the
transformed number {that is, the number as it looks after route pattern or route list digit
manipulation has been applied).
Configure a Translation Pattern for Calls That Are Natively Using Globalized Format
Step 22 Create a translation pattern that is used by the HQ phones with the following
settings:
• Translation Pattern: \+!
• Partition: Global
Note Translation patterns are urgent by default. Make sure that you do not disable Urgent Priority.
• PSTN calling numbers are shown in localized format as described in the activity procedure.
• Callbacks can be placed from \iQ and BR phonesto the PSTN. The calling number for
callbacks is set as described in the activity procedure.
30 Implementing Cisco Unified Communications Manager, Part2 (CIPT2) v8.0 © 2010 Cisco Systems, Inc.
Verify Inbound PSTN Calls to the HQ Site
Place test calls from the PSTN to an HQ phone. Use the local, national, and inlemalional line of
the PSTN phone for the test calls.
Step 15 When placing a call from the local line of the PSTN phone, the caller ID shown on
the display of the HQ phone should be 5554444.
Note You can call the HQ phones by any valid PSTN number, regardless of the line of the PSTN
phone that you use Valid numbers are: 555-2001 and 555-2002, 0-51x-555-2001 and 0-
51x-555-2002, 00-55-51X-555-2001 and 00-55-51X-555-2002.
Step 16 When placing a call from the national line of the PSTN phone, the caller ID shoun
on the display of the HQ phone should be 06065554444.
Step 17 When placing a call from the international line of the PS'fN phone, the caller ID
shown on the display of the I IQ phone should be +776065554444.
Note You can call the BR phone by any valid PSTN number regardless of the line of the PSTN
phone that you use. Valid numbers are: 555-3001, 1-52x-555-3001, and 011 -66-52x-555-
3001
Step 19 When placing a call from the national line of the PSTN phone, the caller ID shown
on ihe display of the BR phone should be 6065554444.
Step 20 When placing a call from the international line of the PSTN phone, the caller ID
shown on the display of the BR phone should be +776065554444.
Verify Callbacks
Step 21 Place callbacks from the IIQ and BR phones by using the entries of the received
calls list. The calls will fail.
Configure Globalization of Calling PSTN Number During Call Ingress at the HQ Gateway
Step 5 Configure the HQ gateway with the following incoming calling party prefixes based
on the number type:
• Unknown: +
• Subscriber: +555U'
• National: +55
• International: +
Configure Globalization of Calling PSTN Number During Call Ingress at the BR Gateway
Step 8 Configure the BR gateway with the following incoming calling party prefixes based
on the number type:
Unknown: +
Subscriber: +665Xv
National: +66
International: +
Step 9 Reset the gateway. Make sure that you also reset the MGCP process at the branch
router by entering the no mgcp command, followed by the mgcp command.
Configure Localization of Calling PSTN Number During Call Egress
Step 10 Create calling party transformation patterns for HQ phones. Refer to the table of the
Task Job Aids.
Step 11 Apply calling party transformation CSS xform-cg_HQ-phones ess to the IIQ
phones. Make sure that you clear the Use Device Pool Calling Party Transformation
CSS check box.
Step 12 Create calling party transformation patterns for BR phones. Refer to the table of the
Task Job Aids.
28 Implementing Cisco Unified Communications Manager. Part 2 (CIPT2) v8.0 ©2010 Cisco Systems, Inc.
— National call: dial any valid national number, for example 9 1 888 666 444. The
National line at the PS'fN phone should ring: the called number that is shown in the
debug output should be 1888666444. and the number type should be unknow n.
— International call: dial any valid international number, for example 9 011 88 222
3.33 4444. The Intemtl line at the PSTN phone should ring; the called number that is
shown in the debug output should be 911882223334444. and the number type
should be international.
This job aid is a\ailable to help you complete the lab task.
Localization of Calling Party During Call Egress for Inbound PSTN Calls
Note At the HQ site, end users expect to see local PSTN callers seven-digit numbers, national
callers with their national numbers and national access codes (0), and international callers
with a + prefix. Because allcaller IDs are globalizedat ingress, there is no need to modify
the caller ID of international callers when sending the call to the phone.
At the BR site, end users expect to see 10-digit caller IDs for local and national callers, and
see international callers with a + prefix (just like at the HQ site).
Activity Procedure
Complete these steps:
Lab Guide
©2010 Cisco Systems Inc
Step19 Reset the BRgateway in Cisco Unified Communications Manager. Make surethat
you alsoresetthe gateway itselfby entering the no mgcpcommand and then the
mgcp command at the BR gateway.
Step 24 Reset the BR gateway in Cisco Unilied Communications Manager. Make sure that
you also reset the gateway itself by entering the no mgcp command and then the
mgcp command at the BR gateway.
Activity Verification
You have completed this task when you attain this result:
• You can place calls to the PSTN from HQ phones. The HQ gateway is used for PSTN
access. Use the debug isdn q931 command for verification of the called and calling
numbers for all types of calls. The calling number should always be seven digits (555 2001
or 555 2002) with number type subscriber. The called number differs per destination:
— Emergency calls: dial 112 and 0 112. The Emergency line at the PSTN phone
should ring; the called number that is shown in the debug output should be 112, and
the number type should be unknown.
— Local call: dial any valid local number, for example, 0 333 4444. Ihe I,ocal line at
the PSTN phone should ring; the called number that is shown in the debug output
should be 3334444, and the number type should be subscriber,
— National call: dial any valid national number, for example, 0 0 888 666 444. The
National line at the PSTN phone should ring; the called number thai is shown in the
debug output should be 888666444, and the number type should be national.
— International call: dial any valid international number, for example 0 00 88 222
333 4444. The Intemtl line at the PSTN phone should ring; the called number that is
shown in the debug output should be 882223334444, and the number type should be
international.
• You can place calls to the PSTN from the BR phone. The BR gateway is used for PSTN
access. Use the debug isdn q93l command for verification of the called and calling
numbers for all types of calls. The calling number should always be 10 digits (5Zv555
3001) with no number type set (unknown). 'Ihe called number differs per destination:
— Emergency calls: dial 911 and 9-911. The Emergency line at Ihe PSTN phone
should ring: the called number that is shown in the debug output should be 911, and
the number type should be unknown.
— Local call: dial any valid local number, for example 9 333 4444. The Local line tit
the PSTN phone should ring: the called number that is shown in the debug output
should be 3334444, and the number type should be unknown.
26 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2)v8.0 © 2010 Cisco Systems, Inc.
Step 4 Save your configuration changes using the copy running-config startup-config
command.
Configure Site-Specific Device Pools and Set the Local Route Group
Step 8 Configure the default device pool with local route group HQ_rg.
Step 9 Create a new device pool that is called BR, and configure the device pool with local
route group BR_rg.
Step 10 Apply the ncuK created device pool BR to the BR phone.
Step 16 Apply called party transformation CSS xform-cd_IIQ-out_css to the IIQ gateway.
Make sure that you clear the Use Device Pool Called Party Transformation CSS
cheek box.
For the BR gateway (MGCP) all digit manipulation is performed in Cisco Unified
Communications Manager. The PSTN expects the BR gateway not to use ISDN number
types; national and international calls have to have the corresponding prefixes.
Localization of Calling Party During Call Egress for Outbound PSTN Calls
Note The calling party number that is sent to the PSTN at the HQ site should use the shortest
possible format (subscriber). The type of number must be set appropriately. The calling
party number that is sent to the PSTN at the BR site should be the 10-digit national number.
The type of number must not to be set.
Activity Procedure
Complete these steps:
Note At this time, all calls that are sent to the HQ-x gateway use G.711 only. However, in the next
lab exercise, TEHO will be enabled. At that time, G.729 will be used for TEHO callers.
Therefore all applicable codecs are configured now.
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 >2010 Cisco Systems, Inc.
Note Only translation pattern 9.911 should be configured with urgent priority. AN other patterns
are unique and therefore urgent priority is not needed. Translation pattern 9.911 could also
match pattern 9[2-9]XXXXXX and is configured with urgent priority so users dialing 9.911
do not have to wait for the interdigit timeout to expire.
Localization of Called Party During Call Egress for Outbound PSTN Calls
\+ l Partition: xform-cd_BR-out
Description: BR (Intl)
Discard Digits Instructions PreDot
Prefix 011
Lab Guide
© 2010 Cisco Systems, Inc
Globalization of Called and Calling Parties During Call Ingress for Outbound
PSTN Calls Placed from BR Phones
22 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 ©2010CiscoSystems, Inc.
Translation Pattern Configuration
These job aids are available to help you complete the lab task.
Globalization of Called and Calling Parties During Call Ingress for Outbound
PSTN Calls Placed from HQ Phones
20 Implementing Cisco Unified Communications Manager. Part 2 (CIPT2) v8.0 ©2010 Cisco Systems, Inc.
Step 5 Enter the following parameters in the Calling Search Space Configuration window:
• Name: HQ-Phones_css
• Description: HQ Phones
Step 6 In the A\ ailable Partitions pane, choose the following partitions and use the down
arrow below the Available Partitions pane to move the highlighted partition to the
Selected Partitions pane. The order of the partitions is not relevant in this lab
exercise:
• Internal
• IIQ_PSTN
Step 7 Click Save.
Step 8 Repeat these steps to add the remaining partitions as they are listed in the "CSSs"
tabic of the Task Job Aids section.
Note Make sure that you assign this CSS to the line level of the phone.
Step 14 from Related Links, chooseConfigure Deviceand click Go to get back lo the
Phone Configuration window.
Step 15 Assign CSS Global_css lo the phone.
Note Make sure that you assign this CSS to the device level of the phone
Step 16 Click Save and then, in the pop-up window, click OK.
Step 17 Repeat these steps to for Phone2-.v.
Step 18 Repeat the above steps for Phone3-.v. butapply CSS RR Phones_css to the fine level
instead of CSS H()_Phoncs_css.
Step 19 Reset all three phones.
Activity Verification
You have completed this task when you attain this result:
• You ha\e created thepartitions thatarelisted inihe"Partitions" table of the task Job Aids
section.
• You ha\e created the CSSs that arc listed in the "CSSs" tabic of the Task Job Aids section.
Lab Guide
) 2010 Cisco Systems. !nc
CSSs
HQ_PSTN
BR_PSTN
Activity Procedure
Complete these steps:
Configure Partitions
In these steps, you will configure the partitions lhat are listed in the "Partitions" table of the Job
Aids section.
Note Make sure to include all partitions as listed in the Job Aids section.
Configure CSSs
In these steps, you will create the CSSs listed in the "CSSs" table of Ihe Task Job Aids section.
Step 4 Na\ igate to Call Routing > Class of Control > Calling Search Space, and click
Add New.
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 ) 2010 Cisco Systems, Inc.
Task 1: Configure Partitions and CSSs
In this task. \ou will configure all partitions and CSSs that arc required for the following tasks.
You will assign partitions and CSSs to phones and phone directory numbers.
These job aids are available to help \ou complete the lab task.
Partitions
Job Aids
Thesejob aids are available to help you complete the lab activity.
Phone Numbers
Intersite Dialing
Phone2-y 8 51/2002
HQ 0 0 00 112,0112
Note For additional information regarding the PSTN, refer to the Dial Plan Information section at
the beginning of this document
16 Implementing Cisco Unrfied Communications Manager, Part 2 (CIPT2) v8.0 >2010 Cisco Systems, Inc.
Lab 1-2: Implementing a Dial Plan for
International Multisite Deployments
Complete this lab activity to practice what you learned in the related module.
Activity Objective
In ibis activity, you will implement a dial plan to support inbound and outbound PSTN calls,
site-code dialing. TFIfO. and PSTN backup. After completing this activity, you will be able to
meet these objectives;
• Configure partitions and CSSs
• Configure inbound PSTN calls
• Configure outbound PSTN calls using H.323 and MGCP gateways
m • Configure site codes tor intercluster calls
• Configure PSTN backup for intercluster calls
• Configure TEHO with local backup
Visual Objective
The figure illustrates what you will accomplish in this activity.
Configure call
routing and digit
manipulation lor a
H 323 PSTN
gateway
Configure call
routing and digil
manipulation at
CUCMI-x for
MGCP PSTN
gateway
Required Resources
These arc the resources and equipment that are required to complete this activity:
• Cisco Unified Communications Manager
• Student PC
• Cisco IP phones
Activity Procedure
Complete these steps:
Add a SIP Trunk in Cisco Unified Communications Manager
Step 1 In Cisco Unified Communications Manager Administration, choose Device >
Trunk.
Step 6 In the SIP Information pane, enter the IP address of the other pod's Cisco Unilied
Communications Manager server: \0.y.\A.
Step 7 Make sure that the Destination Address is an SRV box is not checked.
Step 8 From the SIP Trunk Security Profile drop-down menu, choose Non Secure SIP
Trunk Profile.
Step 9 From the SIP Profile drop-down menu, choose Standard SIP Profile.
Step 10 Click Save, and then, in the pop-up window, click OK.
Step 11 Reset the newly added SIP trunk.
Activity Verification
You have completed this task when you attain these results:
• The SIP trunk appears in the list when you choose Device > Trunk and then click the Find
button in Cisco Unified Communications Manager Administration.
14 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 © 2010 Cisco Syslems. Inc.
Reduce the Utilized B-Channels from 23 to 8
Step 11 Disable the configuration server byentering the following command in global
configuration mode:
no ccm-manager config
Step 12 In global configuration mode, enter the following commands to shut down the voice
port that is associated with the Tl PRI:
voice-port 0/0/0:23
shutdown
Note Because you deactivated the configuration server feature, the MGCP process at the Cisco
IOS gateway is not automatically reset anymore when you reset the gateway in Cisco
Unified Communications Manager. You have to manually reset the MGCP process at the
Cisco IOS gateway every time after you reset the gateway in Cisco Unified Communications
Manager Enter the no mgcp command, followed by the mgcp command, in order to reset
the MGCP process at the Cisco IOS router.
Activity Verification
You have completed this task when you attain this result:
• Your MGCP gateway is successfully registered with Cisco Unified Communications
Manager. This successful registration can be verified at the gateway as follows:
LIsc the show ccm-manager hosts command. The status should show Registered.
— Use the show mgcp endpoint command. All controlled ISDN PRI cndpoinl ports
should be up.
— Use the show mgcp command. The Admin State and ihe Opcr State should be
active.
Use the show isdn status command. Ihe Layer 2 slate should be
MUI.TIPI.FJ:RAMFS F.STABUISIIF.D.
• Verify that the MGCP gateway and the MGCP endpoinls are registered in Cisco Unified
Communications Manager:
Step 1 Navigate lo Device > Gateway.
Step 2 Choose the option lo Shaw endpoinls and click Find. The status of the endpoint
should be registered with IO.v.I.I.
Activity Procedure
Complete these steps:
Log in to BR-x
Step 1 From PC-.v. connectto your headquarters router(HQ-x) using Telnet to
10_v.250.l02. Log in using the password cisco and switchto enable mode(using the
password cisco again).
Note The gateway is currently not configured with any MGCP commands. The T1 or E1 controller
is not configured with a PRI group command. There is no ISDN PRI.
Step 3 Display the network interfaces and their IP configurations by entering Ihe show ip
interface brief command.
Step 4 Display the IP routing table by entering the show ip route command.
Configure the Cisco IOS Gateway for MGCP Using the Configuration Server Method
Step 5 Fnter the terminal monitor command to display the debug output that is generated
by the router.
Step 6 Enter the debug ccm-manager con fig-down load events command to debug the
configuration server feature events.
Step 7 In global configuration mode, enter the following commands:
ccm-manager config server 10.x.1.1
ccm-manager config
Step 8 Monitor the debug output to verify the operation of the configuration server feature.
Turn off all debugging by entering the no debug all command.
Step 9 Fnter the show running-config command. Your gateway should be configured for
MGCP, "fhe configuration that was added by the configuration server feature
includes MGCP, controller, and ISDN PRI settings.
Step 10 Save your configuration changes using ihe copy running-config startup-config
command.
12 Implementing Cisco UnrfiedCommunications Manager, Part 2 (CIPT2) v8.0 ©2010 Cisco Systems, Inc.
Note Further verification will be done in the next lab exercise.
Note These steps are platform-dependent. Cross-check with your instructor to find out the actual
hardware that is used for your MGCP gateway. This lab guide is based on the Cisco 2811
Integrated Services Router platform with a VW1C2-1MFT-T1E1-T1 in slot 0/0. You can also
use the show diag command at your router to display its hardware configuration.
Step 4 [{nterthe following parameters in the Gateway Configuration window, and then
click Save:
Step 6 Click the port icon on the right of the displayed endpoint 0/0/0.
Step 7 In the Gateway Configuration window, enter the following parameters:
• Description: BR-*, Branch PSTN Gateway (MGCP): ISDN PRI
• Device Poo!: Default
CUCMl-y: lO.v.l.l
HQ-y. loopback: 10 v.250.101
HQ-y. voice servers network: lO.y.1.101
HQ-y. phone netwo -k: IO.v.2.101
HQ-y. data network: lO.v.3.101
HQ-y. serial interface to BR-y: I0.y.6.102
HQ-v. serial interfa> e to HQ-x- 10.z.y. 101
BR-y. loopback: I0v.250.102
BR-y. phone network: lO.y.4.102
BR-.v. serial interface to HQ-y: 10.y.6.102
Note As mentioned earlier, x is the number of your pod, and y is the number of your partner pod
(in the same group). Groups are pods 1 and 2, pods 3 and 4, pods 5 and 6, and pods 7 and
8 A z in an IP address stands for the pod number of your pod that is followed by the number
of your partner pod. The z parameter is compounded in ascending order. Examples: for pod
1,x = 1, y = 2, soz = 12; for pod 2, x = 2,y= 1, soz= 12; for pod 3, x= 3. y = 4, soz = 34,
for pod 4, x = 4, y = 3, so z = 34, and so on.
Step 8 Save jour configuration changes using copy running-con tig start up-con fig.
Activity Verification
You have completed this task when you attain these results:
• Verity that H.323 is enabledon the headquarters router on the loopback interlace:
Enter the show running-config interface loopback 0 command to verify the 11.323
interface configuration df the HQ-.r router.
10 Implementing Cisco Unified Communications Mansger, Part 2 (CIPT2) v8.0 ©2010 Cisco Systems, Inc.
Add an H.323 Gateway to Cisco Unified Communications Manager
Step 2 Navigate to Device > Gateway to display the find and List Gateways window.
Step 3 Click the Add New button.
Step 4 from the Gateway '1 ype drop-down menu, choose 11.323 Gateway.
Step 5 Click Next.
Step 6 In the Gateway Configuration window, enter and choose the following parameters:
• Device Name: 10_v.250.101
Note An x in device names or IP addresses stands for your pod number. Ask your instructor if you
are not sure which pod number to use.
Step 7 Leave all other parameters at their default settings, and then click Save.
Step 8 In ihe pop-up window, click OK, and then reset the newly added gateway.
Activity Verification
You have completed this task when you attain these results:
• You ha\e added a new H.323 gateway in Device > Gateway.
Activity Procedure
Complete these steps:
Log in to HQ-x
Step 1 From PC-x connect to your headquarters router (HQ-x) using Telnet to
I0_v.250.l01. If prompted, log in using the password cisco and changelo enable
mode (using the password cisco again).
Step5 Display the IP routing table by entering Ihe show ip route command.
Step 6 Test IP connectivity to the following IP addresses using the ping or trace command:
• CUCMI-x 10-t.I.I
Job Aids
Thesejob aids are available to help you completethe lab activity,
Activity Procedure
Complete these steps:
Implementing Cisco Unified Communications Manager. Part 2 (CIPT2) v8.0 ) 2010 Cisco Systems, Inc.
Lab 1-1: Implementing Basic Multisite
Connections
Complete this lab activity to practice what you learned in the related module.
Activity Objective
In ihis activily. you will implement an H.323 gateway and an MGCP gateway for connecting to
the PSI'N and a SIP trunk for connecting to the other pod. AHercompleting this activity, you
will be able to meet these objectives:
Add an H.323 gateway to Cisco Unified Communications Manager
Configure an H.323 gateway
Add an MGCP gateway to Cisco Unified Communications Manager
Configure an MGCP gateway
Configure a SIP trunk in Cisco Unified Communications Manager
Visual Objective
'] he figure illustrates what you will accomplish in this activity.
Required Resources
fhese are the resources and equipment that arc required lo complete this activity:
• Cisco Unified Communications Manager
• Student PC
:; »;xx x iiS5..XXX
b [2^:X< [i."•';>* xtxx ;;-?jxx \: U.^.-J55-3XXX
00-55-51 <-55S-£XX>;
G 'nTf' [2 u'XX *Xi* 55S-3XXX
'i :.ffi p.-j;xx <xx< 0-«i-555.3XXX
00*9 Ml-5 5.1-3 XXX
.Tf P'l
5!. 5 2 XXX
i [2 y)y.t 1 51iE5!)?XXX
(HIMMlBSSiXXX
1 80C [J • 5SXXXX 5553XXX
1 i'OO [2- <X*XXX 1 52* 555 3XXX
011 SB 521 555 3XXX
y Code: 66
5553XXX 555 3XXX
52I5553XXX j Access Code: 9 52/555 3X XX
66 521 555 3XXX Ial Access Code: 1 66 52y 555 3XXX
nal Access Code: 011
Note The blue text boxes towards PSTN-Phone-x indicate how thenumber isdialed by HQ-x and
BR-x users (leftblue box£s)
;s) and how the number has to be sent to the PSTN on the PRI
(right blue boxes)
The orange text boxes tctwards the HQ-x and BR-x sites indicate how the number is dialed
by the PSTN user. The fll ure does not show how the numbers are delivered to the HQ-x
and BR-x gateways on It 3 PRI.
Implementing Cisco Unified Communications Manager. Part 2 (CIPT2) vfl.O 12010 Cisco Systems, Inc.
As shown in the table, thecalling number that isused by the PSTN phone depends on the
PSf N phone line that is used to place the call.
800 "PSTN"
Note When calls are placed from line 800, the calling number is not provided, the calling name
"PSTN" is presented instead of a calling number.
Controller type E1 T1
international calls (E.164 number of any length), 011 {E.164 number of Intemtl
TON = international any length)
Note The PSTN at the HQ sites allows remote calling numbers to be sent (for example in case of
TEHO or device mobility) The calling number that is sent to the PSTN through HQ
gateways should always be in the shortest-possible format. Calls originating at the local HQ
site should use local format, calls originating at the HQ site of the other pod should use
national format, and calls originating at one of the two BR sites should use international
format
The PSTN at the BR sites does not allow remote calling numbers to be sent. The calling
number that is sent to the PSTN through BR gateways should always be in national format.
Calls originating at any remote site (BR site of the other pod, one of the two HQ sites)
should use the number of the local BR attendant (52x-555-3001).
Calls to the HQ and BR sites can be placed from PSTN-Phone-*, as shown in the table:
Note The table shows the valid numbers that can be dialed at PSTN-Phone-x. No PSTN access
code is dialed from the PSTN phone. The presentation of the called number for these calls is
different The called number is always presented to the HQ and BR gateways without
prefixes and the TON is always set.
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 ) 2010 Cisco Systems, Inc
The table pro\ ides an overview of the dial plan that is in use.
PSTN range in international format 55 51* 555 2XXX 66 52x 555 3XXX
The PSTN accepts calls from the HQ and BR sites as shown in the table.
International calls (E 164 number of any length), 011 (E.164 number of any
TON = international length)
Note When HQ gateways are sending calls to the PSTN, the gateways are required to set the
ISDN TON for the called number (see the table for details); national and international
prefixes must not be used.
When BR gateways are sending calls to the PSTN, the gateways are required to send
prefixes for national and international calls; the TON must not be set (it must be set to
Unknown).
The used PSTN numbenng plans do not fully represent the actual numbering plans that are
used in North America or Europe They only use some components of these PSTN
numbering plans.
Ifa dialed number matches the range that is used at one of the HQ or BR sites, the PSI'N routes
the call to that sile. All other valid calls are sent lo PSIN-Phone-.*. as shown in the table:
BR_y Ptione3-y
10 y.250.102/3 4_ffiL_
1011 0/24 iA~-* (PSI^? 10jj4 0/24 T*'
*-_* M02 DHCP
Pody
Ihe x in the figure indicates your pod number. They in the figure indicatesthe numberof the
pod that will work together with you.
Implementing Cisco Unrfied Communications Manager. Part 2 (CIPT2) v8.0 © 2010 Cisco Systems. Inc.
CIPT2
Lab Guide
Overview
1 his guide presents the instructions and other infonnation concerning the lab activities for this
course. You can find the solutions in the lab activity Answer Key.
Outline
This guide includes these activities:
Lab 1-1 Implementing Basic Multisite Connections
Lab 1-2 Implementinga Dial Plan for Internalional Multisite Deployments
lab 2-1 Implementing SRST and MGCP fallback
Lab 2-2 Implementing Cisco Unified Communications Manager Kxpress in SRSI" Mode
Lab 3-1 Implementing Bandwidth Management
Lab 3-2 Implementing CAC
Lab 4-1 Implemenling Device Mobility
Lab 4-2 Implemenling Cisco Fxtension Mobility
Lab 5-1 Implementing Cisco SAP and CCD
Answer Ke\
Lab 3-2: Implementing CAC 67
ActivityObjective 67
Visual Objective 67
Required Resources 67
Job Aids 68
Task 1: Configure Locations 68
Task 2: Configure RSVP-Enabled Locations 69
Task 3: Configure AAR and CFNB to Route Calls over the PSTN If They Are Not Admitted by the
Deployed CAC Methods 72
Task4 (Optional): Configure SIP Pre[x>nditions 73
Lab 4-1: Implementing Device Mobility 75
Activity Objective 75
Visual Objective 75
Required Resources 75
Task 1: Configure Device Mobility 76
Lab 4-2: Implementing CiscoExtension lability 79
Activity Objective 79
Visual Objective 79
Required Resources 79
Task 1: Activate the Cisco Extension Mobility Service and Configure the Corresponding Service
Parameters 80
Task 2: Create a Device Profile for a User 81
Task 3: Add and Associate an End U ser with the User Device Profile 82
Task 4: Add the Cisco Extension Mollility IP Phone Service and Subscribe to IP Phones and
Device Profiles 83
Lab 5-1: Implementing Cisco SAF and CI ;D 85
ActivityObjective 85
Visual Objective 85
Required Resources 85
Task 1: Configure SAF Forwarder Fu ictionality on the HQ-x and BR-x Router 86
Task 2: Configure Cisco Unified Corr nunications Manager as SAF Client 87
Task 3: Configure Cisco Unified Corr nunications Manager Express SRST on Branch Router to
Leam Routes Using CCD 91
Answer Key 96
Lab 1-1 Answer Key: Implementing Basic Multisite Connections 96
Lab 1-2 Answer Key: Implementing a Dial Plan for International Multisite Deployments 96
Lab 2-1 Answer Key: Implementing RST and MGCP Fallback 96
Lab 2-2Answer Key: Implementing (Jisco Unified Communications Manager Express in SRST
Mode 96
Lab 3-1 Answer Key: Implementing Ejandwidth Management 96
Lab 3-2 Answer Key: Implementing CAC 96
Lab 4-1 Answer Key: Implementing Cevice Mobility 96
Lab 4-2 Answer Key: Implementing Cisco Extension Mobility 96
Lab 5-1 Answer Key: Implementing . AF and CCD 96
Implementing Cisco Unified Communications Manager. Part 2 (CIPT2) v8.0 ) 2010 Cisco Systems, Inc.
Table of Contents
Lab Guide
Overview 1
Outline 1
Lab Topology 2
Dial Plan Information 2
Lab 1-1: Implementing Basic Multisite Connections 7
Activity Objective 7
Visual Objective 7
Required Resources 7
Job Aids 8
Task 1: Add an H.323 Gateway to Cisco Unified Communications Manager 8
Task 2: Configure an H.323 Gateway 9
Task 3: Add an MGCP Gateway to Cisco Unified Communications Manager 11
Task 4: Configure an MGCP Gateway 12
Task 5: Configure a SIP Trunk in Cisco Unified Communications Manager 14
Lab 1-2: Implementing a Dial Plan for International Multisite Deployments 15
Activity Objective 15
Visual Objective 15
Required Resources 15
Job Aids 16
Task 1: Configure Partitions and CSSs 17
Task 2: Configure Outbound PSTN Calls 20
Task 3: Configure Inbound PSTN Calls 27
Task 4: Configure Intersite Calling 31
Task 5: implement TEHO Within Your Pod 35
Task 6: Implement TEHO Between Pods 37
Lab 2-1: Implementing SRST and MGCP Fallback 39
Activity Objective 39
Visual Objective 39
Required Resources 39
Task 1: Configure SRST Gateways in Cisco Unified Communications Manager 40
Task 2: Configure a Cisco IOS Gateway for MGCP Fallback and SRST 40
Task 3: Implement a Dial Plan in Cisco Unified Communications Manager Supporting Outbound
Calls During SRST Mode 42
Task 4: Implement a Dial Plan at the SRST Gateway Supporting Inbound and Outbound Calls
When in MGCP Fallback or in SRST Mode or Both 43
Lab 2-2: Implementing Cisco Unified Communications Manager Express in SRST Mode 47
ActivityObjective 47
Visual Objective 47
Required Resources 47
Task 1: Configure Cisco Unified Communications Manager Express in SRST Fallback Mode 48
Task 2: Configure MOHon Cisco Unified Communications Manager Express 49
Lab 3-1: Implementing Bandwidth Management 51
Activity Objective 51
Visual Objective 51
Required Resources 51
Job Aids 52
Task 1: Enable Software Media Resources on Cisco Unified Communications Manager 53
Task 2: Configure Regions 54
Task 3: Implement Transcoders 57
Task 4: Implement a Hardware Conference Bridge 58
Task 5: Implement Multicast MOHfrom Branch Router Flash 61
5-74 Impiementing Cisco Unified Communications Manaj :f, Part 2(CIPT2) v8.0 >2010 Cisco Systems, Inc.
Module Self-Check Answer Key
OD D.G
y_) h.L)
04) The toDID ruledescribes how to manipulate thelearned UN pattern in order to gelto thenumber h.
should be used for a PSTN backup call if the CCD path is unavailable.
(.)>) i:
06) D
5-72 Implementing Cisco Unitied Communications Managt -, Part 2 (CIPT2) vB.O 12010 Cisco Systems, Inc.
Module Self-Check
Use the questions here to re\ iew what \ou learned in this module, fhe correct answers and
solutions are found in the Module Self-Check Answer Key.
Ql) Which two ofthe following devices do not support CCD? (Choose two.) (Source:
Implementing SAF and CCD)
A) Cisco Unified SRST
B) Cisco IOS gateway
C) Cisco Unified Border Element
I)) Cisco IOS gatekeeper
F.) Cisco Unified Communications Manager
F) Cisco Unified Communications Manager Express
G) Cisco IOS Catalyst switches
Q2) Which two statements are true about SAF? (Choose two.) (Source: Implementing SAF
and CCD)
A) SAF forwarders interpret the SAF header and SAF service data.
B) An internal SAF client is allocated with a SAF forwarder.
C) An internal SAF client resides in Cisco Unified Communications Manager.
D| SAF clients do not have to be Layer2-adjaccnt.
1.) SAF requires FIGRP tobeused as the IProuting protocol.
Q3) Which two statements arc not true about CCD? (Choose two.) (Source: Implemenling
SAF and CCD)
A) Call routing information is learned by the CCD requesting service.
B) Call routing infonnation is advertised by the CCD advertising service.
C) Load balancing occurs among trunk protocols and learned remote IP addresses.
D) Learned call routing information can be placed into different partitions that are
based on the remote call control identity.
F) Learned call routing information can be placed into different partitions that are
based on the remote IP address.
Q4) What is the purpose of the toDID rule in CCD? (Source: Implementing SAF and CCD)
Q5) Which of the following is not aconfiguration step when implementing SAF in Cisco
Unified Communications Manager? (Source: Implementing SAF and CCD)
A) Configure SAF forwarder.
B) Configure SAF trunk.
C) Configure CCD advertising and requesting service.
D) Configure hosted DN group and hosted DN pattern.
F) Configure DN blockprofile.
F) Configure blocked learned patterns.
Module Summary
This module started with a description of Call Control Discovery (CCD). which is a feature that
allows callagents to advertise andlearn dial plan information to and from a CiscoService
Advertisement Framework (SAF)-enabled network. It showed how SAF" works, how CCD
utilizes SAF. and how SAF and CCD are implemented in a Cisco Unified Communications
solution.
References
For additional infonnation. refer lo these resources:
• Cisco S\ stems. Inc. Cisco Unified Communications System 8.x SR\D. April 2010.
hup:. \sww.cisco.eom/cti/US/docs/voice ip_eomm/euem/snid/S*/ue8\.html
• Cisco Systems. Inc. Cisco Unified Communications Manager Administration Guide
Release 8.0/1). February 2010.
lmp:;www.eisco.com.'en.'LS/docs/\'oicc ip comni/cucm/admin/8 OLccnicfg/hccm-801
cm.html
Summary
References
For additional infonnation. refer to tlicse resources:
Cisco IOS Service Advertisemen Framework Configuration Guide 15.1 at
http://www .cisco. com/en/US/dt /ios/saf/configura£ion/guide/saf_cg _ps 105')2_TSD_Prod
uctsC'onfiguration GuidcC haft':er.html.
Cisco Systems. Inc. Cisco Unifiid
Uniji Communications System 8.x SRND, April 2010.
htlp:'V\\w\\.cisco.(jom/en/US.'''do s/voicejp eomm/cuem/sntd/8xAic8\.himl
CiscoSystems. Inc. Cisco Unifi
ifiei Communications Manager Administration Guide
Release 8.0(1). February 2010.
http://w ww.cisco.com/en.T ;S/doc s/voice_ip_comin/eucm/admin/8_0 l/ccmcig/hccin-801-
cm.html
Implementing Cisco Unified Communications Manage -. Part 2(CIPT2) v8.0 »2010 Cisco Systems, Inc.
Other SAF and CCD Considerations
This subtopic describes additional issues that you need to consider when implementing CCD in
Cisco Unified Communications Manager.
Ifvou do not assign atrunk when you configure the CCD requesting service. Cisco Unified
Communications Manager uill not subscribe to the SAF forwarder. No routes will be learned.
Each hosted DN pattern must be globally unique.
Ifatrunk is assigned to aroute group or is associated with aroute pattern, you cannot enable
SAF on the trunk, and vice \ ersa.
You cannot enable SAF on SIP trunks that use authenticated or encrypted security profiles.
When you are configuring aCisco Unified Communications Manager cluster with one ormore
SAF forwarders, by default, all Cisao Unified Communications Manager nodes that are applied
to the SAF-enabled trunk or trunks Via the device pool will register with the configured SAF
forwarder.
First, you have to make sure that each local node uses adifferent SAF client ID. You can easily
check the nodes by using a SAF client name that ends with @. In this case, each node will
utilize the configured name that is followed by @and aunique node ID. At the SAF forwarder,
you have toadd the basename keyword tothe end ofthe SAF external-client client-ID
command, or you have to manually ^onfigure all names that are used by the nodes in your
cluster.
In some cases,you may not want all nodes that should use SAF register with all configured
SAF forwarders. For example, as shbwn in the figure, when youuse clustering overthe WAN,
you typically want to registernodes only with their local SAF forwarders. For that
configuration, click the Show Advanced link at the SAF forwarder configuration page.
At the advanced configuration mode page, you can associate individual members of the cluster
selectively with the configured SAF forwarders.
5-66 Implementing Cisco Unified Communications Managbr, Part 2(CIPT2) v8.0 12010 Cisco Systems. Inc.
Trunk Considerations When Using Globalized Call Routing
This subtopic describes considerations regarding the different trunk types when using
globalized call routing.
As mentioned earlier, vou can configure one SAF-enabled SIP trunk or one SAf-enabled H.32.^
trunk in Cisco Unified Communications Manager and in Cisco IOS Software. II both types of
trunk are used bv the advertising SAF client and both are used at the requesting SAF client,
then all routes are learned twice—once per protocol. When placing acall to such aroute, load-
sharing will occur, This means that half of the calls are setup using SIP, and half of the calls
are signaled by H.323.
When implementing TEHO and using globalized call routing, TFHO calls are expected to be
received with a- prefix. The reason is that they are advertised that way and the incoming VoIP
call can be routed back out to the PSTN at the TEHO gateway when the called number is in
globalized format.
II 323 trunks however, do not send the +sign. When acall is received (or placed) through an
H.323 trunk and the called number includes a+. the +sign is stripped. This does not happen on
SIP trunks.
When vou rclv on the +to be received through the H.323 trunk, you have to configure
incoming called-partv settings at the 11.323 trunk. Consequently, the +is prefixed before the
received called-party number is matched in the call-routing table without the +.
If vou have aSIP and an 11.323 trunk, and you do not prefix the 4- at the H.323 trunk due to the
load-sharine algorithm, even' second call would fail (H.323) while the other halfofthe calls
would work (SIP). Such apparently inconsistent errors are difficult lo troubleshoot.
Note When you expect to receive VoIP calls to internal directory numbers as well as to globalized
(PSTN) numbers, make sure that your incoming called-party settings prefix only the +to the
called numbers where it is required. You can either refer to the ISDN type of number or use
global transformations in order to control which called-party numbers you can modify.
TEHO Considerations
1TEHO destinations are located atthe PSTN and do not exist intemally.
• Advertised hosted DNis PSTNdestinationand not internal DN.
IfSAF-leamed TEHO route becomes unavailable, ToDID number is usedfor
automatic backup call.
- Typically, both numbers are identical (ToDID rule is 0:) anduse E 164format
with + prefix.
TEHO destinations are located at tfye PSTN only; they do not exist internally at all. The
advertised directors' number is aP^TN number. When globalized call routing is enabled, this
number has to be in E. 164 format vvith a + prefix.
Calls toPSTN destinations will match the learned directory' number and are therefore sent to
the TEHO site over the IP WAN. lit case the IP WAN is down, the local gateway should be
used as a backup. This situation reduires having aToDID rule of0:. With this rule, the CCD
PSTN backup number is identical tp the learned directory number. When the IP path is marked
as unreachable, the same number would be called using the AAR CSS ofthe calling phone.
In Cisco Unified Communications Manager, generate aToDID rule of0: by checking the Use
Hosted DN as PSTN Failover check box. In Cisco IOS Software, you cannot set the ToDID to
Further, ifglobalized call routing is to be used, you are forced to use global patterns, which do
not allow any ToDID rule to be advertised.
When TEHO pattern is advertised without aToDID rule, local TEHO backup does not work
You could only configure static local backup routes by putting similar patterns into partitions
that are listed later in the phone CSS. Ilowever. such patterns are used only after the learned
pattern has been purged completely. By default, this process occurs after the expiration of the
CC DPSTN Failover Duration timer, which is 48 hours by default.
Based on these issues, it is recommended that you not advertise TEHO patterns from Cisco IOS
Software ifthe + is required and local backup is desired.
gg.
5-64
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 »2010 Cisco Systems, Inc.
Solution for PSTN Backup Advertised in E.164 Format Without
Leading +
['his subtopic describes the workaround for CCD PS'fN backup calls lo PSTN destinations that
are in E.164 format without a •+ prefix.
It would be vcrv cumbersome to add all possible E. 164 numbers lo the overall dial plan of
Cisco Unified Communications Manager. However, because CCD PSTN backup calls are
always placed with the use of the AAR CSS of the calling phone, you can add one translation
pattern ' to apartition that is accessible only from the AAR CSS. At that pattern, you can prefix
a+to the called-party number and set the CSS of the translation pattern to aCSS that has
access to the global PS'fN route pattern (\+!).
That process solves the issue with CCD PS'fN backup calls. Ilowcver. ifAAR is enabled, it
will break AAR. assuming that the AAR implementation is based on globalized call routing. IS
the external phone number mask of the destination phone is in E. 164 format with a+prefix,
then AAR calls would not work anvmore. The reason is that they use the same CSS and
therefore would also match the !translation pattern that prefixes a+. In this case AAR calls
would be placed to E. 164 numbers with two +signs. In order to also make AAR CSS cal s
work vou have to add asecond translation pattern into the same partition that is accessible
from the AAR CSS onh -This second translation pattern \+! is not configured with any digit
manipulation but uses the same CSS as the other translation pattern (!). As aconsequence.
AAR calls are passed on to the V+! route pattern without any digit manipulation (by matching
the more specific \+! translation pattern, which docs not prefix a+). CCD PS 1Nbackup calls
do not match the Vi-! translation pattern and are therefore routes, as explained earlier.
Note
The described solution is a workaround only. The implementation of advertised patterns in
Cisco IOS Software may be changed in the future so that +can be configured in the
advertised pattern and in the ToDID rule. If so, you should change from the described
workaround to the solution that allows CCD PSTN backup calls to be placed to globalized
numbers. .
p-atlla da-tlock 1 i l l u -
P-.-~_ 197.555 4XXX
• The+ not supported In-lias-prefix or extension.
p«e-«B l tj-» uiuuios 0:1972555 • PSTN backup canrroi be advertised with +.
Cisco IOS internal SAF clients have limited support regarding the +sign in advertised routes.
In fact, the +sign cannot be configured in either the directory number pattern or the ToDID
rule, "fhe only way to advertise apattern with +is to use the pattern tag type global command
instead the pattern tag type extension command. In this case, however, Ihe ToDID is always
unset, regardless ofthe configured alias prefix at the directory number block profile.
When aCCD-enabled Cisco Unified Communications Manager uses globalized call routing for
PST Naccess, the mentioned limitation ofCisco IOS internal SAF clients causes issues because
the backup PSTN number is not in a format that Cisco Unified Communications Manaeer can
route to the PSTN.
The workaround is to make sure that CCD PSTN backup calls can be routed to the PSTN even
if the number that results from the ToDID rule does notstart with the+.
Note
Cisco IOS Software has limitations only in advertising patterns that include the +sign. Cisco
IOS Software can process received patterns that include a +sign without any problems or
limitations.
5-62
Implemenling Cisco Unified Communications Manager, Part 2(CIPT2) V8.0 ©2010 Cisco Systems, Inc.
CCD and Static Routing Integration Considerations
This subtopic describes how CCD can be integrated with static routing in Cisco Unified
Communications Manager.
All routes that are learned by CCDare put intothe same partition.
If partition islisted first in CSS. il has pnority tor equally qualified matches
Partition allows learned routes tolake precedence over statically configured
backup routes.
Make surethatbackup routes in later partitions are not more specific than
learned hosted DNs
Routes in later partition areconsidered only after learned entry iscompletely
deleted
• The learned IP path istried until CCD Learned Pattern IPReachable
Durationexpiration (default is 60 seconds)
IftheIPpathdoesnot work during this time, thecall fails.
• ToDID is usedas backup after expiration ofCCD Learned Pattern IP
Reachable Duration until expiration ofCCD PSTN Failover Duration (default
is 48 hours)
IfnoToDID is configured, the callfailsduring this time.
• The learned pattern iscompletely removed only after expiration ofCCD
PSTN Failover Duration.
Static backup patterns are now considered.
All routes that are learned bv CCD are put into the same configurable partition. Ifthis partition
is listed first in the CSS ofthe calling phones, ithas higher priority for equally qualified
matches than partitions that arelisted later.
Such aconfiguration allows learned routes to take precedence over statically configured backup
routes You have lo make sure that backup routes in later partitions are not more specific than
learned routes, because the order ofpartitions is relevant only ifthe matches are equally
qualified.
Be aware that routes in later partitions arc considered only after learned routes are removed
from the call-routing table.
When Cisco Unified Communications Manager loses IP connectivity to its SAF forwarder, it
wails for 60 seconds until it considers the IP path to be unavailable. You can configure this
time by using the CCD learned Pattern 11' Reachable Duration CCD feature parameter. During
that time, and calls to learned patterns fail.
Once the timer has expired. Cisco Unified Communications Manager starts another timer, the
CCD PSTN Failover Duration. The default value for this timer is 48 hours. During this time
Cisco Unified Communications Manager tries to place aCCD PSTN backup call. 11 no loDID
has been advertised. Cisco Unified Communications Manager assumes that there is no ISIN
backup path and that therefore calls will fail.
The learned route is purged only after the expiration of the timer. Then another (statically
configured backup) pattern, which is in apartition that is listed after the CO) partition, can be
matched If vol, want to use locally configured static backup patterns, etther disable t CO
PSTN backup b> setting the CCD PSTN Failover Duration timer to 0. or set the timer to a
lower value than the default {twodays).
Hfc
5-60 Implementing Cisco Unified Communications Manager, Pari 2(CIPT2) vB.O 12010 Cisco Systems, Inc.
SRST Considerations
This subtopic describes how CCD can be implemented at Cisco Unified SRST gateways.
SRST Considerations
ACisco Unified SRST gateway does not need to advertise any internal directory numbers
because the SRST site is reachable only via the PSTN. It is the responsibility ofCisco Unified
Communications Manager to know how to route calls lo cluster-internal directorv' numbers
when they arenot reachable overthe IP WAN.
However the Cisco Unified SRST gateway needs a local dial plan that allows end users to
place calls to other sites b\ dialing the internal directory number ofthe other site. Cisco linified
SRS r then must transform the internally used directory numbers to the corresponding PS1N
numbers so that the call can be rerouted over the PSTN. This local dial plan does not have to be
configured manually when CCD is used. Instead, Cisco Unified SRST can subscribe to SAf
and hence learn all internallv known DN ranges and the corresponding ToDID rules. Cisco
Unified SRST learns these routes while there is no network problem. At this time, the learned
patterns are not utilized because the Cisco Unified SRST gateway does not route any calls:
Cisco Unified Communications Manager is in control ofall IP phones and performs call-
routing sen, ices.
Once IP connectivity is broken. IP phones fall back to the Cisco Unified SRST gateway, and
once roistered, the Cisco Unified SRST gateway has to route calls. Because the gateway has
learned all available internallv used directory numbers with the corresponding ToDID rules, it
can now route to the respective PSTN number any calls that are based on the dialed internal
directory number.
In the example, the Cisco Unified SRST gateway learned three patterns while IP connectivity
was working: 8408XXXX with aToDID rule of4:+l408555, 8415XXXX with aToDID rule
of4:+l8415. and 8949XXXX with a ToDID rule of4:+l949222.
When a learned pattern is marked unreachable anda ToDID has been advertised withthe
pattern,a PSTN backup call is placed. The CSS that is used for this call is the AAR CSS.
Make sure that the AAR CSS is set atall phones, so that PS'fN backup calls for CCD-learned
patterns will work. Also ensure that the number that is composed ofthe directory number
pattern and the ToDID rule isroutable (in other words, that a route pattern that matches the
number exists).
Note PSTN backup for CCD iscompletely independent from AAR. AAR isused toplace PSTN
backup calls for cluster-internal destinations when the IP path cannot be used because of
insufficient bandwidth as indicated by CAC.
It is only the AAR CSS that is reused for CCD PSTN backup. Otherwise, CCD PSTN backup
does not interact with AAR at all. For example, CCD PSTN backup works even when AAR is
globally disabled by the corresponding Cisco CallManager service parameter.
5-58 Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 !> 2010 Cisco Systems, Inc.
Monitoring Learned Routes in Cisco Unified Communications
Manager Express
"Ihis subtopic describes how to monitor learned routes in Cisco linified Communications
Manager Express.
Pattern - 2XXX
Primary Trunk-Bouts(B> ID : 273 274
Aliaa-Routels) Prefix/Strip-Lan : -.498952121/0
Pattern - 3XXX
Primary Trunk-Route(s) ID : 270 269
Alias-Route(B) Pretix/Strip-Len : ♦44228822/0
Pattern - *4420>
Trunk-Route!-} ID •- 271 272
Pattern - .16505051234
Trunk-Fouta(g) ID : 270 269
The figure shows how to display the SAF-leamed routes in Cisco IOS Software. Note that onl_
acall agent can interpret SAF service data; SAF forwarders cannot interpret SAF service data.
Therefore, this command works only on Cisco IOS routers that arc internal SAF clients as well
as SAF forwarder.
The output shows two types ofpatterns: extensions (with aToDID) that were learned from a
de\ ice other than aCisco IOS device, and global patterns, which include a+sign and no
ToDID infonnation (most likely advertised by another Cisco IOS internal client).
Oump-n
iM-rt-N. ctnuw-i -,
JraSBf„ a*» .Eatm _-j»<*
'3HMJ1118563) Reich**
nitMi(fflio)
!408707131; ?3f»3n!iS«3] Rotfuc* H323 tnunci?:oi
2matw2W()j:7 RtmaM SlttidWnKMlr 0151t(SM(i
SUnrnkiwCluslH 015 IH3J143f t>-umm
M(lW)ni1SSS3J HM!hj»
oiuiiaowj tSWKW
;ji«oiriis»ij rium
01132 tci?.ai
331*0*1185(3; Rsjuuw Stan-UcneriLisiBi 01511(5060)
_H1OTJM1J1» 35 Reichit* HSM SWkWWwCIusIb. ai.tici.t43t
SAF-leamed routes are not visible by any tool of the Cisco Unified Communications Manager
Administration web page. The only way to view SAF-leamed routes is by using the Cisco
Unified Real-Time Monitoring Tool (RTMT).
The figure shows an example ofSAF-leamed routes that are displayed by Cisco Unified
RTMT. The ToDID rule 0: means that the '-internal" pattern and the pattern that is used for
PSTN backup are the same pattern. This principle usually applies when advertising TEHO
patterns are advertised. The ToDID rules that are empty mean that there is no PSTN backup
path forthe respective learned patterns.
5-56
Implemenling Cisco Unified Communications Manager, Part 2<CIPT2) v8.0 >2010 Cisco Systems, Inc.
CCD Considerations
This topic describes important issues that you need lo consider when implementing CCD and
SAF.
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prof 11a trunk-rout*
••••Ion protc • lp intarfaca loopbmckl tr.ui.port top pot
iflla da-block 1 allaa-pratlx 1972555
P» 1 typa «t«n»ioQ 4111
icollli c.llt
da-jarvlca
trunk.tout*
da-block 1
The configuration ofthis dial peer is like the configuration ofadial peer that refers to asession
target ras command when an H.323 gatekeeper isused. The destination pattern .Tstands for
all learned routes. The rest ofthe dial peer configuration is used as atemplate for the outgoing
dial peer that is irsed on outbound SAF calls, and for the incoming dial peer that is used on
inbound SAF calls.
Ifyou have other dial peers that also represent learned routes, the preference command will be
used to detemiine which dial peer should be treated with higher priority.
5-54
Implementing Cisco Unified Communications Manager. Part 2(CIPT2) vB 0 >2010Cisco Systems, Inc.
Step 5: Configure Requesting Service
"fhe figure shows the configuration of the advertising service in Cisco IOS Sotlware.
si-interface PastEtherriStO/0
topology base
exit-sf-topology
exit-service-faaily
profile trunk-route 1
Bession protocol sip interface loopbackl transport tcp port 5060
profile dn-block 1 alias-prefix 1972555
pattern 1 type extension 4XXX
I
profile callcontrol 1
trunk-route 1
dn-block 1
You can also configure the requesting service under channel tag vrouter EiGRP-ID asystcm
-IS Use the subscribe callcontrol wildcarded command to enable the learning ofroutes that
are advertised by the SAF process that matches the autonomous system number that is specified
at the channel configuration level.
ervice saf
pro fi e trunk-route 1
rofile callcontrol 1
Actual SAF client, which you can enable
dn-service foradvertising (shown here)and learning
trunk-route 1 (shownin nextfigure). Advertising service
dn-block 1 (publish) refers to call control profile.
channel 1 vrouter SAF asyste Refers to EIGRP process and
publish callcontrol 1
autonomous system number.
You configure the advertising and requesting services under channel tag vrouter E/GRP-ID
asystem AS. "fhe EIGRP-ID argument refers to the name that was assigned to the router EIGRP
process (SAF. in the example shown). The AS argument is the autonomous system number that
wasassigned to the EIGRP service family.
To enable the advertising service itself, you use the command publish callcontrol tag The tag
argument refers to the tag that was applied to the previously configured call control profile
Effectively, you configure the call control profile (which determines which directory numbers
should be advertised by which trunk protocol) by the SAF process that is identified by the
autonomous svstem number.
5-52 Implementing Cisco Unifier) Communications Manager, Part 2(CIPT2) vS.O © 20)0 CiscoSystems, Inc.
Step 3: Configure Call Control Profile
The figure shows the configuration ofthe call control profile in Cisco IPS Software
sf-interface FastEthernet0/0
topology base
exit-sf-topology
exit-service-family
fhe call control profile refers to one or more directory number blocks and to atrunk. 1he call
control profile will be used in the next step to specify that the listed directory number blocks
should be advertised at the specified trunk or trunks (if two ofthem are used). Another
command that xou can enter under dn-smice is site-code site-code extension-length length. It
allows asite code to be prefixed to all configured extensions referenced by the call control
profile The extension-length argument sets the number ofdigits (starting with the least
significant digit) that should be preserved from the configured extension before the site code is
added.
sf-lnterface FastEthernet0/0
topology base
exit-sf-topology
exit-service-family
1
Each directory number block is configured globally with aToDID that is applied to all
extensions that are listed later. The command to configure adirectory number block is profile
dn-block tag alias ToDID-prefix strip ToDID-strip. The subsequent command toadd
extensions is pattern lagtype extension pattern.
Note The ToDID-strip argument stands for the number of digits to be stripped; the ToDID-prefix
argument stands for the prefix tobeadded tothe internal number after stripping digits.
Neither the ToDID-prefix argument nor the pattern argument support the use of the +sign. If
you want to advertise anumber with a+sign, you have to use the command pattern tag type
global pattern. Again, you cannot enter the +sign in the pattern argument; however, due to the
type global, a+sign is prefixed to the configured pattern. The ToDID ofglobal patterns is
always unset.
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Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 i* 2010 Cisco Systems, Inc.
Step 1: Configure Trunk Profile
The figure shows the configuration ofatrunk profile in Cisco IOS Software.
The trunk profile is con figured with the interface that should be used for call signaling. It is
configured also with the protocol type (in this case. SIP) and the transport parameters (I CP
versus UDP. and port number).
You can configure one SIP trunk or one 11.323 trunk.
The configuration steps that are listed in the figure are steps thai you can do multiple times, if
multiple SAF forwarder processes are configured in separate autonomous systems. Each SAF
client channel that is configured with the advertising and the requesting service has to refer to
anotherSAF autonomous system.
The CCD advertising service ofasingle SAF client channel can refer to multiple call control
profiles. This capability allows the configuration oftwo trunk profiles (one SIP and one H.323
trunk per call control profile). Only one trunk isrequired.
5-48 Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 S2010 Cisco Systems. Inc.
CCD PSTN Failover Duration: This parameter specifies the number ofminutes that calls
that are placed to learned patterns that have been marked unreachable are routed through
PSTN faikner and are then purged from the system. For the duration that isspecified in
this parameter to start counting down, another service parameter, CCD Learned Pattern IP
Reachable Duration must first have expired. Theexpiration of thatparameter indicates that
IP connectivih is down between the SAF forwarder andCisco Unified Communications
Manager, and that all learned patterns are marked unreachable. Then, when the duration in
this parameter. CCD PSI'N Failover Duration, expires, all learned patterns are purged from
the system. Also, calls to purged patterns are rejected (the caller hears a reorder tone or a
"This number is unavailable" announcement). Setting this parameter lo 0 means that PSTN
failover is disabled. If the SAF forwarder cannot be reached for the number of seconds
defined inthe CCD Learned Pattern IP Reachable Duration service parameter, and no
failo\er options are provided through the PSTN, then calls to learned patterns will
immediateK fail. Setting this parameter to525600 means that PSTN failover will never
expire and.'as aresult, learned patterns will never be purged due to loss of communication
with the SAF forwarder, "fhe default is 2880 minutes (48 hours).
Issue Alarm for Duplicate Learned Patterns: This parameter determines whether Cisco
Unified Communications Manager issues an alarm called DuplicateLearnedPattern when it
learns duplicate patterns from different remote call control entities on the SAF network.
The default value is False.
(CD Stop Routing On Unallocated I nassigncd Number: ihis parameter determine*
whether Cisco Unified Communications Manager continues to route calls to Ihe next
learned call control entitv (ifadvertised by multiple call agents) when the current cail
control entitv rejects the call with the cause code for Unallocated/Unassigned Number. An
unallocated number represents a hosted directory number that does not exist in the current
call control entit\ . The default value is True.
fli" ^s*«a
® **• ****!
••n-ajva.
Remote IP
CCD blocked learned patterns are optional. If CCD blocked learned patterns are configured, all
routes that match any of the configured criteria arc blocked. As aresult, they are not added to
the call-routing table.
You can configure afilter that is applied to received routes in order to deny the learning of
routes, using these criteria:
• [earned pattern: The received pattern is checked in its entire length. If it matches ihe
configured learned pattern, it will not be added to the local call-routing table.
• 1earned pattern prefix: The received patterns arc compared with the configured prefix,
sorting with the left-most digit. By using alearned pattern prefix for blocking received
routes you can filter internally used numbers by their leading digits-tor example, by their
site code.
. Remote call control identity: Each call agent has aso-called SAF client ID By setting the
remote call control identity, you can filter received routes that are based on the ID of the
ad\ertising call agent.
• Remote IP: B> setting this filler, you can block routes that arc based on the advertising IP
address.
You can configure only one CCD requesting service. You have to enter the partition that all
learned routes should be put into. You must first create the partition as shown in the figure.
In addition to creating the partition, you can configure the CCD requesting service with a
learned pattern prefix and aPSTN prefix. These prefixes are applied to all learned DN patterns
and to all learned ToDID rules, respectively.
Finally, the CCD that isrequesting service isreferred tothe SAF-enabled SIP or to the SAF-
enabled H.323 trunk.
Ifyou associate the CCD requesting service with only one type oftrunk, all received routes that
are reachable by the other (unconfigured) protocol type are ignored. They are not added to the
call-routing table.
5-44 Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 ;>2010 Cisco Systems, Inc.
Step 6: Configure CCD Advertising Service
The figure shows the configuration of the CCV> advertising service in Cisco Unified
Communications Manager.
SVHIH*™-.. , p
You need to configure one CCD advertising service for each eonligured hosted DN group. ^
bach CCD advertising sen ice can use the SAF-enabled SIP trunk or the SAF-enabled H.323
trunk. One trunk has to be specified. Multiple CCD advertising (and the CCD requesting
sen ice)can refer to the same SAF-enablcd trunks.
®'-
- Hotted ON PMttra rrrfo •
Ho««fl »*t»re '
DeiciDbon
Hosted im Group"
Hosted DN patterns refer to ahosted DN group. As mentioned earlier, ifthe parameters at the
hosted DN pattern are unset, the parameters ofthe hosted DN group are applied. When the
PSTN Failover Strip Digits field is set to 0and the PSTN Failover Prepend Digits field is
empty, both fields are considered tobe unset. The configuration example that isshown in the
figure does not generate aToDID rule of 0:, but it applies the settings of the configured hosted
DN group.
At the hosted DN group, the same logic applies. Ifthe PSTN Failover Strip Digits field is set to
0and the PSTN Failover Prepend Digits field is empty at the hosted DN pattern and at the
hosted DN group, then the no ToDID rule is advertised. As aresult, there is no PSTN backup
when the IP path is unavailable.
If you want to advertise aToDID rule of 0:, the number that should be used for backup is
identical to the internally used number (for example, when TEHO patterns are advertised)
Therefore, you have to check the Use Hosted DN as PSTN Failover check box.
®-
The ToDID rules used for
- ltDflcd DN Sr
PSTN backup are applied to
hosted DN patterns only if not
Tconfigured at DNpattern
05™ ItJuvtK SOB DIOJIS 0
The hosted DN group will be referenced from hosted DN patterns. Ifall—or at least most—of
the associated hosted DN patterns share the same ToDID rules, you can configure the ToDID
rule at the hosted DN group. The settings of the hosted DN group arc applied to the hosted DN
patterns if the hosted DN pattern parameters areunset.
The Use Hosted DN as PS'fN Failover check box instructs Cisco Unified Communications
Manager to create aToDID rule of 0:. As aresult, the number that is to be used for PSTN
backup is identical to the internally used number. Usually, this result occurs only when tail-end
hop-off (TF. HO) patterns are advertised.
-- -"- - ' in
S!>T.ur* I
[*™ tow'
You can configure one SAF-enabled SIP trunk (as shown in the figure) and one SAF-enabled
H.323 trunk. With aSAF-enabled H.323 trunk, you have to first add astandard nongatekeeper-
controlled ICT and then check the Fnable SAF check box. Once the check box ischecked, the
IP address field is disabled. The reason is that the configured trunk does not refer to aparticular
destination IP address but instead acts as a template for adynamically created trunk once a SAF
call isplaced. The destination IPaddress isthen taken from the learned SAF service data.
The same concept applies to the SAF-enabled SIP trunk. The only difference is that the SAF-
enabled SIP trunk is a special trunk service type, which isselected before the trunk
configuration page isshown. Therefore, there isno extra check box like there is atthe
nongatekeeper-controlled ICT. The SAF-enabled SIP trunk also does not have adestination IP
address field.
5-40
Implementing Cisco Unified Communications Manager, Part 2{CIPT2) v8.0 ©2010 Cisco Systems, Inc.
Step 2: Configure SAF Forwarder
The figure shows how to add aSAF forwarder to Cisco Unified Communications Manager.
The destination IP address has to match the one ofthe interface that is specified with the sf-
interfaee command at the SAF forwarder.
If you want to register with more than one SAF forwarder, click the Show Advanced link. This
link allows you to configure multiple SAF forwarders and to associate individual members oi
the cluster selectively with the configured SAF forwarders.
Note If you want to allow multiple nodes of aCisco Unified Communications Manager clusier to
act asSAF clients, each of them needs a unique client name. You can either configt re each
of them individually with separate node names or use aSAF client ID in Cisco Unified
Communications Manager, which is ctient-ID@ The @sign instructs Cisco Unified
Communications Manager to add a unique node number sothat the actual client IDs are
client-ID@1, client-ID@2, and so on.
At the SAF forwarder, you can either create individual entries or add the keyword basename
to the external-client client-ID command. Do not specify the @sign at the SAF forwarder:
only add the keyword basename to the external-client command, and the specifieo client
ID will bepermitted with any suffixes of @followed by a number.
©«
This must match the usemame
and password configured at the
(trqitiw.
SAF forwarder.
Make sure that the username and the password match the username and password that were
configured at the SAF forwarder.
5-38 implementing Cisco Unified Communications Manager, Pari 2(CIPT2) v8.0 >2010 Cisco Systems, Inc.
External SAF Client Configuration Procedure
This subtopic shows the configuration procedure ofan external SAF client.
"fhe last two configuration steps are optional. You do not have to configure the CCD
advertising sen ice and the CCD requesting service ifyou want only lo advertise or leam call
routes (exclusively).
Fach allowed exiemal client has to be listed in the service-family section. In addition, the
username and password that should be used by the external client have to be specified in the
service-family external client section.
Note If you want to allow multiple nodes of a Cisco Unified Communications Manager cluster to
act asSAF clients, each of them needs aunique client name. You can either configure each
ofthem individually with separate node names oruse a SAF client ID in Cisco Unified
Communications Manager, which isclient-ID@. The @sign instructs Cisco Unified
Communications Manager to add a unique node number sothat the actual client IDs are
client-ID®-}, client-ID@2, and so on.
At the SAF forwarder, you can either create individual entries or add the keyword basename
to the external-client client-ID command. Do not specify the @sign atthe SAF forwarder;
only add the keyword basename to the external-client command, and the specified client
ID wjjl bepermitted with any suffixes of@followed by a number.
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Implementing Cisco Unifiea Communications Manager, Part 2(CIPT2) v8.0 32010 Cisco Systems, Inc.
Step 1: Configure SAF Forwarder
he figure shows the SAF forwarder configuration.
sf-interface FastEthernetO/0
Autonomous system must be
topology base
the same on all SAF forwarders
exit-sf-topology
that should exchange data
exit-service-family
"~l
In the example, aSAF forwarder is configured with autonomous system I. All SAF forwarders
that should exchange information with each other have to be in the same autonomous system.
You use the sf-interface command to bind the SAF process to the specified interface. Ifthe
router has multiple interfaces, if is recommended that you use aloopback interface.
The configuration ofthe SAF forwarder consists oftwo steps: the SAF forwarder configuration
(mandatory) and the support ofan external SAF client (if used).
5-34 Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 ©2010 Cisco Systems, Inc.
Relationships of Internal SAF Client Configuration Elements
This subtopic describes the relationships ofinternal SAF client configuration elements.
SAF Client
DN Block DN Block
SAF CH«« "Charmer Profile Profile
CCD Recjueslmg
Service
Call Control Profile
CCD Advertising
Service
Trunk Profile
The figure illustrates how internal SAF client configuration elements relate to each other.
Note The configuration that is shown in the figure is one that you can do multiple times, if multiple
SAF forwarder processes are configured in separate autonomous systems. Each SAF client
"channel" has to refer to another SAF autonomous system
The CCD advertising service of a single SAF client channel can refer to multiple call control
profiles This capability allows the configuration of two trunk profiles (one SIP and one H.323
trunk per call control profile). Only one trunk isrequired.
Configuration
Element Name Configuration Element Function
The table shows the configuration elements ofan internal SAF client, their functions, and the
ways that they interact with each other.
Note You can configure the advertising service and the requesting service independently of each
other.
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Implementing Cisco Unified Communicalions Manager, Part 2(CIPT2) v8.0 >2010 Cisco Systems. Inc.
Relationship of External SAF Client Configuration Elements
This subtopic describes the relationship ofexternal SAF client configuration elements.
The figure illustrates how external SAF client configuration elements relate to each other.
Note
You can configure only a single CCD requesting service in your cluster. You can configure
mu
Itiple blocked learned patterns, SAF forwarders, and SAF security profiles. You a n
co
nfigure one CCD SIP trunk and one CCD H.323 trunk. Only one trunk is required
SAFtrur*s
One SAF trunk andone SAF H323 twr* canbeconfigured. Ttiey arenot
configure
configured' Hti
ti a destination IPaddress.Therest ofthe confaurafjon isstrraiar
to normals 3andH323r/uni».
Nested DN group Configured nthi PSTN laitover stripdigrtsand PSTN failover prepend diaits
1 DN patterns
CCD requesting service Configured<iitti route partition, teamed pattern prefix, and PSTN prefix. Refers to
Blocked learned patterns Configurediith remote IP. remote call control identity, and learned pattern or
learned pre! ;
The table showsthe configuration elements of anexternal SAF client, their functions, and the
ways that they interact with each o :her.
«*
5-30
Implementing Cisco Unified Communications Mana jer, Part 2(CIPT2) v8.0 >2010 Cisco Syslems, Inc.
High-Level Configuration Overview (Cont)
Implement external SAF client:
• Add external SAF client (Cisco Unified Communications Manager) to SAF
forwarder:
Specify SAF ID, username, and password of external client.
Map external SAF client toSAF autonomous system.
• Add SAF forwarder (Cisco IOS router) toSAF client (Cisco Unified
Communications Manager).
- Specify SAF ID. username, and password asconfigured at SAF
forwarder.
• Configure CCD at external SAF client.
Configure SAF trunks
- Configure hosted DN patterns andhosted DN groups.
ConfigureCCD advertisingservice.
Configure CCD requesting service and partition for learned patterns.
When implementing external SAF clients, you must perfonn these high-level configuration
tasks:
Stepl At the SAF forwarder (Cisco IOS router), add the external SAF client (Cisco
Unified Communications Manager):
• Specif\ the SAF ID. username. and password ofthe external client.
• Map the external SAF client to the SAF autonomous system.
Step 2 At the external SAF client (Cisco Unified Communications Manager), add the SAF
forwarder (Cisco IOS router):
• Specify the SAF ID. username. and password as configured at the SAI
forwarder.
- Enable requesting se
ConfigureVoIP dial peer eferring to SAF.
The first main configuration task : ;to enable SAF inthe network. You have toconfigure SAF
forwarder functionality on a Cisoc IOS router. All SAF forwarders must share the same SAF
autonomous system number. You :an specify the interface thatshould beused bytheSAF
forwarder.
Whenusing internal SAF clients, •ou mustperform these main configuration tasks:
Step 1 Configure a trunk prof e and specify the interface and theprotocol to beused for
call signaling.
Note The IP address (interfaci) that is used for call signaling can be different from the IP address
that is used by theSAF forwarder.
5-28
Implementing Cisco Unified Communications Mans per, Part 2(CIPT2) v8.0 12010 Cisco Systems, Inc.
Note if the leamed pattern was removed when the IP path became unavailable, the originating
site would not know what PSTN number to use for the backup call. By default, a route is
completely removed only if it has not advertised for 48hours.
BR Learned Routes
When a user at the HQ site dials I00I during the link failure at the BRsite, the called numberis
still found in the call-routing tabl of the HQ Cisco Unified Communications Manager cluster.
However, the IP path is marked i ireachable, and therefore the call cannot besetup over the IP
network with the use of a SIP tru ik
Cisco Unified Communications M anager now checks whether there is a ToDID rule that is
associated with the leamed patte i. In this case, a ToDID rule of 0:+1972555 has been learned.
Cisco Unified Communications Manager does not strip any digits {because ofthe 0 in front of
the : [column]), but adds the pref +1972555 to the dialeddirectorynumber 4001. The
resulting number +19725554001 s now matched in the call-routing table of Cisco Unified
Communications Manager, when a match is found in a PSTN routepattern.
In the example, the ToDID rule results in globalized PSTN numbers (E.164 format with a +
prefix) Therefore, a PSTN route pattern that matches this format (for example, \+.!) has to
bein place. If all sites share the same PSTN dial rules-for example, all sites arewithin the
North American Numbering Plan (NANP)—then you could also configure ToDID rules that
result in PSTN patterns with a PSTN accesscode, followed by a national accesscode
followed by the 10-digit PSTN number. In this case, your PSTN route pattern would have to
be91[2-9]XX[2-9]XXXXXX.
5-26 Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 © 2010 Cisco Systems, Inc.
CCD—Link Failure at BR
This subtopic describes how CCD manages a link failure between the SAF client and its SAF
forwarder at the BR site.
CCD—Link Failure at
•- - PSTN-
•49S953i2ixxxx>s:;:::;;
HQ 101510
SAF-Enabled
IP Network
When the connection between the SAF client and the SAF forwarder at the BR site isbroken,
the SAF forwarder atthe BR site detects this problem that is based on the missing keepalives of
the registered SAF client.
The BR SAF forwarder sends an update throughout the SAF-enabled network so that all SAF
forwarders are aware that the IP path to4XXX is currently unavailable. Other than in IP
routing, the learned route is not removed, but only the IP path is marked unreachable.
All SAF forwarders that have registered SAF clients now pass this update on totheir SAF
clients, so that all SAF clients in the network can mark the IP path lo4XXX as unreachable.
As shown in the figure, the call-routing table at the HQ site also gets updated accordingly.
CCD—Call from HQ to BR
BR Learned Routes
PSTN;
The HQ site dials 4001. The called number is found in the call-routing table ofIhe IIQ Cist
Unified Communications Managercluster.
Note All learned routes are put into the same configurable partition The CSS of the calling phone
must have access to this partition in order for the call towork. If the calling phone does not
have access to the partition that includes all learned patterns and there isno match in any
other partition, the call fails.
5-24 Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 >2010 Cisco Systems, Inc.
CCD—Propagation of BR Routes
This subtopic describes how BR routes are propagated to the IIQ site.
CCD—Propagation of BR Routes
W&nmmmWmMmW mmnV^&mmmmmn^R Mf
SAF-Enabled
IP Network
The Cisco Unified Communications Manager cluster atthe BR site advertises its direeto >
number range 4XXX with aToDID rule of0:+1972555 to its SAF forwarder, fhe SAF
network propagates this new route throughout the network, and the SAF forwarder at the HQ
site sends the information to the HQ Cisco Unified Communications Manager cluster. The call-
routing table of the HQ cluster is populated with the directory number pattern 4XXX and a
ToDID rule of 0:^1972555.
Again, onh aSIP trunk has been associated with the CCD advertising service at the originating
site. Therefore, the IIQ cluster learns the route only forSll\
The network is in a converged state. All sites know about the routes ofall other sites.
CCD—Propagation of HQ Routes
HQ Learned Routes BR Learned Routes
97255SXXXX
Advertise hosted DN
range (2XXX) and ToDID
rule (0:^-498953121
The Cisco Unified Communications Manager cluster at the HQ site advertises its directory
number range 2XXX with a ToDID rule of0:+49895312I to itsSAF forwarder. The SAF
network propagates this new route throughout the network, and the SAF forwarder atthe BR
site sends the information to the BR Cisco Unified Communications Manager cluster. The call-
routing table ofthe BR cluster ispopulated with the directory number pattern 2XXX and a
ToDID rule of 0:+498953121.
At the advertising site, only aSIP trunk has been associated with the CCD advertising service.
Therefore, the BR cluster learns the route only for SIP.
5-22
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 >2010 Cisco Systems. Inc.
CCD Operation
This topic describes how CCD works for on-net calls and how CCD reroutes calls to the PSTN
if the IPpath is notavailable.
CCD—Base Configuration
The figure shows the base configuration. There are two sites, each with aCisco Unified
Communications Manager cluster. One site (••HQ" in the figure) is located in Germany, and it
has -i DID ran°e of +4989^3121 XXXX. Intemally, range 2XXX is used. The other site ("BR )
is in the United States: it has a DID range of+ I972555XXXX. Internally, the directory number
range 4XXX is used.
m
Note
PSTN backup for CCD is completely independent from AAR. AAR is used to place PSTN
backup calls for cluster-internal destinations when the IP path cannot beused because of
insufficient bandwidth as indicated by Call Admission Control (CAC).
It is only the AAR CSS that is reused for CCD PSTN backup. Otherwise, CCD PSTN backup
does not interact with AAR atall. For example, CCD PSTN backup works even when AAR is
globally disabled by the corresponding Cisco CallManager service parameter.
5-20 Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 ©2010 Cisco Systems. Inc.
Processing Received Routes in Cisco Unified Communications
Manager
This subtopic describes how received routes are processed in Cisco Unified Communications
Manager,
You can configure afilter that is applied to received routes in order to deny the learning of
routes, using these criteria:
• Iearned pattern prefix: The received patterns are compared with the configured prefix,
starting with the left-most digit. By using alearned pattern prefix for blocking receiv ed
routes, you can filter intemally used numbers by their leading digits—lor example, by their
site code.
• 1earned pattern: The received pattern is checked in its entire length. If it matches the
configured learned pattern, it will not be added to the local call-routing table.
• Remote call control identity: Each call agent has aso-called SAI-' client ID. By setting the
remote call control identity, you can filter received routes that are based on the ID o. the
advertising call agent,
. Remote IP: By setting this filler, you can block routes that are based on the advertising IP
address.
You can configure one or more criteria when setting up afilter. However, as soon as one
criterion ismatched, the learned route is filtered.
The same destination number can be learned multiple times. It may be advertised by different
call agents. It mav allow SIP and 11.323 to be used for setting up the call (both signa ing
protocol capabilities are advertised separately). It also may be reachable at multiple IP
addresses (of the same call agent, in the case of aCisco Unified Communications Manager
cluster). Ifa route is learned multiple times, Cisco Unified Communications Manager™! load-
share the outbound calls to the corresponding destination among all possible paths (that is. t>>
protocol and remote IP addresses).
Call Control Discovery
S 2010 Cisco Systems. Inc
Note Like thetrunks that areassociated with CCD advertising services, thetrunks thatare
associated with theCCD thatis requesting services are not used to learn patterns via SIP or
H323. They determine theoutbound capabilities for calls thatare placed tolearned
destinations. If the CCD that isrequesting service isassociated only with an H.323 trunk,
learned routes that areto bereached via SIP are not added to the call-routing table of the
receiving Cisco Unified Communications Manager.
The Cisco Unified Communications Manager nodes of the cluster that ispermitted to place
outbound calls to learned routes are determined by the device pool that isapplied to the
trunk that is associated with the CCD requesting service.
mm
5-18 Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 >2010 Cisco Systems, Inc.
CCD Services in Cisco Unified Communications Manager
This subtopic describes the two CCD services that exist in acall agent such as Cisco Unified
Communications Manager.
fhe CCD advertising sen ice is configured with the directory' numbers that are to be advertised.
In Cisco Unified Communications Manager, they are configured by so-called hosted DN
(directorv number) ranges. Fach hosted DN range is configured with its PSTN failover
information (the ToDID rule for the hosted DN range). In addition, the signaling protocol and
the IP addresses ofthe call agents have to be advertised. They are configured by a trunk. The
trunk can be a SAF-enabled H.323 intercluster trunk (ICT) ora SAF CCD SIP trunk. CCD
advertises call routes with one ormore call agent IP addresses. 1he IP addresses tobe
advertised are determined bv the device pool that isapplied tothe SAF-enabled trunk.
Note The trunk is not used to advertise call routes Call routes are advertised by CCD and SAF
and not via H323 orSIP. The trunk is used todetermine the IP addresses ofthe call agents
and the supported signaling protocols, in case another call agent wants to establish a call to
a learned call route
The CCD that is requesting service is responsible for subscribing to call-routing infonnation
from its SAF forwarder. It allows Cisco Unified Communications Manager to leam routes from
the SAF-enabled network. Onlv one CCD that is requesting service exists per Cisco Unified
Communications Manager cluster. However, like the advertising service, it can be configured
to accept patterns that are reachable via SIP or H.323. depending on the associated trunk or
trunks.
In the given example, the complete ToDID rule would be 4:+1408555, because the numbers to
be stripped and the prefix are separated by a column.
By advertising only the locally present internal numbers and the corresponding ToDID rule at
each call agent, the dial plan implementation oflarge networks is extremely simplified. Ifthere
are any changes at acall agent, you have to change only the advertised number (range) and its
ToDID mle at the affected call agent. All other call agents will dynamically leam the changes
5-16
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 >2010 Cisco Systems, Inc.
CCD Characteristics
Ihis subtopic explains the main characteristics ofCCD.
CCD enables call agents to exchange call-routing information. The infonnation that is relevant
consists of these components:
• Dial plan information: Dial-plan infonnation includes internally used directory numbers
(potentially with internal prefixes such as site codes), the IP addresses ofthe respective call
agents and the signaling protocol that will be used by the call agents. Ali ofthis
information is advertised by call agents that is propagated throughout the network by SAP.
and then learned by other call agents.
• Reachability information: This dynamic routing for call reachability infonnation
drastically simplifies dial plan implementations in large networks. There is no need lor a
static full-mesh configuration and no need even for the configuration of acentralized call-
routing sen-ice (such as an H.323 gatekeeper or aSIP network service). You have tc
configure onlv the internal number range that should be advertised per call agent at the
respective call agent. CCD and SAF then ensure that the locally known numbers are
distributed among all call agents.
When rerouting over the PSTN is desired, call agents are configured not only to ^crtise their
internallv used number ranges, but also with the corresponding PSTN numbers, Hie PS 1N
number is not advertised as adistinct number, but it is advertised by aPMNtailovcr digit
transformation rule that is known as aToDID rule. AToDID rule describes how he mtemally
used number has to be manipulated to get to the associated PS TN number. AIoDID rul.
consists of two components:
CCD Overview
CCD is a function ofcall agents. Itallows call agents to advertise locally known internal
directory numbers and the corresponding PSTN numbers to other CCD-enabled call agents.
CCD utilizes SAF for distributing call-routing infonnation over the SAF-enabled network.
ACCD-enabled call agent is configured to send its locally configured directory number range
as a SAF service to a SAF forwarder. The CCD SAF client generates SAF service data (call
reachability information) and passes iton to the SAF forwarder that will propagate the
information within the SAF network. All SAF forwarders that have SAF subscribing clients
that are attached send the SAF service data to their clients. From aCCD perspective, all SAF
clients exchange SAF service data (call-routing information).
You can compare the SAF service data with the TCP or the User Datagram Protocol (UDP).
which establishes an end-to-end communication between IP endpoints. Likewise, CCD-enabled
call agents exchange call-routing infomiation via the end-to-end service data exchange.
The SAF header can be compared to the IP header. It is also interpreted at intermediate nodes
(SAF forwarders), while these intermediate network nodes do not process the payload (that is.
the SAF service data).
5-u Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 © 2010 Cisco Systems, Inc.
SAF Client and SAF Forwarder Functions
This subtopic describes SAF client and SAF forwarder functions.
SAF clients register to the network, more precisely lo aSAF forwarder. They can publish
sen ices (that is. advertise infonnation) to the SAF network or subscribe to services (that is.
request infonnation) from the SAF network. In order to allow the SAF client and the SAF
forwarder to quickly detect dead peers (for example, ifthe device was powered off), they
exchange keepalives.
SAF forwarders propagate updates that are received from SAF clients that publish sen ices to
other SAF forwarders. Thev send updates to SAF clients, which subscribe to services. Ir.
addition. SAF forwarders exchange hellos with other SAF forwarders in order to detect dead
peers.
SAF forwarders can be. but do not have to be, directly neighboring devices. If they are Layer 2
adjacent, there are two configuration options:
• Multicast: When multiple SAF forwarders are connected via abroadcast-capable medium
like a LAN using Fthernet, they can communicate toeach other via multicasts. This
communication allows adynamic neighbor discovery because there is no need to statically
configure the Layer 2 adjacent neighbors.
• Unicast: When it is not desired that all SAF devices on abroadcast-capable medium
automatically discover each other. SAF forwarders can be configured to send updates only
to statically configured neighbors via unicast messages. In the figure, the lower-left
example shows three routers that are connected to an Fthemct. However, other than in the
upper-right example, they should not build adjacencies among each other in a full-mesh
fashion. Instead, they should communicate only in ahub-and-spoke fashion (one router
communicates with both ofthe others, and the other two routers do not communicate
directly with each other).
When SAF forwarders are not Layer 2adjacent—that is, when there are one or more IP hops
between them—these nonadjacent neighbors have to be statically configured. No discoverv is
possible. See the Hlustration in the lower-right comer of the figure for an example of non-L aver
2 adjacent SAF forwarders.
5-12
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8 0 H2010 Cisco Systems. Inc.
SAF Routing Characteristics
This subtopic describes the main characteristics ofSAF routing.
- Split horizon
Authenticated updates
- Incremental updates (only when changes occur)
• Independent of IP routing protocol
Static
• Dynamic (EIGRP. OSPF, BGP, and so on)
The SAF-FP uses features and functions of the Enhanced Interior Gateway Routing Protocol
(F1GRP) for SAF routing. Features and mechanisms that are utilized and known from EIGRP
include the DifTusing Update Algorithm (DUAL) to prevent loops, reliable transport over IP (IP
protocol 88). support for authenticated updates, and incremental, event-triggered updates tor
fast convergence and low -bandwidth consumption. Configurable parameters that relate to these
FIGRP-derived features include bandwidth percent, hello interval, holdtime. split horizon,
maximum hops, andmetric weights.
Although SAF routing is verv like EIGRP, it is independent of the used IP routing protocol,
SAF works over static routing, as well as in networks that use dynamic routing protocols such
as FIGRP Open Shortest Path First (OSPF). and Border Gateway Protocol (BGP).
5-10 ImplementingCisco Unified Communications Manager, Part 2 (CIPT2)v8.0 )2010 Cisco Systems, Inc.
SAF Client Types
This subtopic describes the two available CCD SAF client types.
A SAF forwarder is alwavs a Cisco IOS router. Remember that SAF forwarders do not process
the propagated service infomiation. Their function is to propagate the information within the
SAF network and lo pass it on to SAF clients.
SAF clients then interpret the service infomiation. With CCD. the SAF client is a call control
device, which sends and receives call-routing infomiation. Depending on the type of call
control device, the CCD device can be an internal or external SAF client:
*• • External SAF client: The SAF client and the SAF forwarder are two different devices.
They use the SAK-C1' for communication. An exampleof a CCD externalSAF client is
Cisco Unified Communications Manager.
• Internal SAF client: The SAF client and the SAF forwarder arc two different functions
within the same device—a Cisco IOS router. They use an internal application programming
interface (API) for communication. Examples of CCD internal SAF clients are Cisco
Unified Border Element. Cisco Unified SRST. Cisco Unified Communications Manager
Express, and Cisco IOS gateways.
SAF Components
Cisco Unified Cisco Unified
SAF supports any service
lo be advertised Communications Communications
Manager Manager
CCD is the first Cisco application
using SAF to advertise services
(call routing)
SAF network components.
• Exchange service
information among
each other
SAF forwarders can interact with SAF clients. A SAF client is an entity thatprocesses SAF
service data.A SAF client can independently advertise (generate) SAF service information to
be propagated in the network, or subscribe to (receive) SAF service information. A SAF client
communicates with oneor more SAF forwarders by the SAF Client Protocol (SAF-CP). With
CCD. the SAF client is a call agent such as Cisco Unified BorderElement, Cisco Unified
SRST. Cisco Unified Communications Manager, Cisco Unified Communications Manager
Express, and Cisco IOS gateways.
5-8 Implementing Cisco Unified Communications Manager, Part2 (CIPT2) vfl.O ) 2010 Cisco Systems, Inc.
Call Control Discovery Overview
This subtopic provides an overview of CCD.
CCD Overview
With CCD.each CCD-enabled call agent advertises locally found directory numbers or
director.' number ranges and their corresponding PS'FN numbers or prefixes to theSAF-
enabled network. In addition, each CCD-enabled call agent learnscall-routing infonnation from
the network.
SAF isused to propagate infomiation within the SAF-enabled network. SAF forwarders
interact with CCD-enabled call agents (thatis. SAF clients). A SAF forwarder learns
infonnation from a SAF client. SAF forwarders exchange learned call-routing information with
each otherso that the SAF-enabled network is aware of all learned call routes. SAF forwarders
do not onh leam from SAF clients, but they alsoadvertise all learned information lo SAF
clients, fhat way. all SAF clients are aware of all available call-routing information—internal
director, numbers and their corresponding PSTN numbers.
The problem of dynamically distributing reachability information is known also in areas other
than call routing. In IP networks, for example, routing has changed from simple static routing
to large, fully dynamic clouds, such as the Internet.
The solution for scalable IP routing is provided by dynamic routing protocols. IP routers have
local networks that are attached. They advertise these locally known networks to other routers
so that all routers can leam about all available networks and the path to get to those networks.
The same concept can be used to distribute call-routing infonnation. Each call-routing domain
advertises locally known telephone numbers or number ranges. Because local numbers are
typically used by internal patterns (using VoIP) as well as via the PSTN, each call-routing
domain advertises both the internally used numbers and the corresponding external PSTN
numbers.
Cisco CCD. a new feature that was introduced with Cisco Unified Communications Manager
Version 8. provides exactly such a service. It allows Cisco Unified Border Element. Cisco
Unified SRST. Cisco Unified Communications Manager, Cisco Unified Communications
Manager Express, and Cisco IOS gateways to advertise and leam call-routing information in
the form of internal directory numbers and PSTN numbers or prefixes.
5-6 Implementing Cisco Unrfied Communications Manager, Part 2 (CIPT2) v8.0 ©2010 Cisco Systems, Inc.
Dial Plan Scalability Issues in Large Networks
The main scalability issues in targe networks are caused by the fact that call-routing
infonnation has to be configured separately at each call-routing domain.
Without centralized services (such as H.323 gatekeepers or SIP network services), a full-mesh
configuration is required. In other words, each call control domain has to be configured with
call-routing information toward all other call-routing domains. This implementation model does
not scale at all and therefore is suitable only for smaller deployments.
In a hub-and-spoke deplovment model, call-routing information for each call-routing domain is
eonligured onk once at the centralized call-routing entity. This eentrali/ed call-routing entity
can be a SIP network service or an H.323 gatekeeper. Such a solution scales better than full-
mesh topologies: however, it introduces a single point of failure and therefore requires
redundant deployment of the centralized service. In addition, the centralized call routing still
has to be manually configured. For example, if telephone number ranges or prefixes are
changed at one of the call-routing domains, these changes also have to be manually performed
at the centralized call-routing service. Further, PSTN backup has to be implemented
independently at each call-routing domain.
In summary, there is no dynamic exchange of call-routing infomiation between call-routing
domains, and there is no automatic PSTN backup.
mm
i 2013 Cisco Systems, Inc. Call Control Discovery
SAF and CCD Overview
This topic prov ides an overviewabout SAFand CCD.
Call Agent
Call Agem
In large networks with several call agents—such as Cisco Unified Communications Manager
Express. Cisco Unified Communications Manager, Cisco Unified Border Element, Cisco
Unified SRST. and Cisco IOS gateways—the implementation and maintenance of dial plans
can be very complex.
The use of 11.323 gatekeepers or Session Initiation Protocol (SIP) network services reduces the
complexity. However, dial plan implementation still does not scale well in very large
deployments.
5^ Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 J2010 Cisco Systems, Inc.
Lesson 1
Objectives
Upon completing ihis lesson, you will beable to describe and implement SAF clients and
forwarders in an environment with CCD. This ability includes being able to meet these
objectives:
Describe what SAF is. what CCD is. and how CCD utilizes SAF
Module Objectives
Upon completing this module, you will be able lo describe and implement CCD deployments.
I his ability includes being able to meet this objective:
• Describe and implement the SAF client and forwarder in an environment with CCD
Module Self-Check Answer Key
QU B
Q2) C. D.G
03) A.C
Q4) C
Q5) B
06) B
Q7) A
Q8) CD
09) A. D. E
Q10) A, D
Oil) A, B
4-74 Implementing Cisco Unitied Communications Manager. Part 2 (CIPT2) v8.0 © 20t0 Cisco Systems, Inc.
Oil) Which two statements describe the result when theuser logs into a device but is still
logged in to another device? (Choose two.) (Source: Implementing Cisco Extension
Mobility)
A) If the multiple login behaviorserviceparameter is set to disallowed, the login
fails.
B) If the multiple login behaviorservice parameter is set lo auto-logout, the user is
automatically logged out of the other device.
C) If the multiple login behaviorenterprise parameter is set to allowed, the login
succeeds and the user also remains logged in at the other device.
D) If the multiple login behaviorenterprise parameter is set lo prompt, the user is
given the option to log out of the other device first.
E) fhe login fails.
2010 C sco Systems, inc. Implementation of Features and Applications for Multisite Deployments
Q6) If no physical location is configured at the device pool, the physical location that is
configured at the phone is used. (Source: Implementing Device Mobility)
A) true
B) false
Q7) Which of the following is not a problem when users roam between sites? (Source:
Implementing Cisco Extension Mobility)
A) The phones that they use have the wrong location and region settings.
B) Users get the wrong extensions on their phones.
C) Users get the wrong calling privileges.
D) Users do not have speed dials available.
Q8) Which two settings cannot be updated when Cisco Extension Mobility is used?
(Choose two.) (Source: Implementing Cisco Extension Mobility)
A) phone button template
B) softkey template
C) device CSS
D) network locale
E) phone service subscriptions
F) phone lines and speed dials
Q9) Which three configuration elements are not relevant for Cisco Extension Mobility
configuration? (Choose three.) (Source: Implementing Cisco Extension Mobility)
A) location
B) phone
C) end user
D) device security profile
F) device pool
F) device profile
Ci) phone service
010) Which two of the following are recommended approaches to implementation of calling
privileges when Fxtension Mobility is used? (Choose two.) (Source: Implementing
Cisco Extension Mobility)
A) Configure the line or lines of the device profile of the user with a CSS that
includes blocked route patterns for the destinations that the user should not be
allowed to dial.
B) Do not configure a device CSS.
C) Do not configure a line CSS.
D) Configure the device with a CSS that includes all PSTN route patterns pointing
to die local gateway.
E) Configure the line or lines of the physical phone with a CSS that includes
blocked route patterns for the destinations that the user should not be allowed
to diai.
4-72 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 © 2010 Cisco Systems, Inc.
Module Self-Check
Use the questions here to review what you learned in this module. The correct answers and
solutions are found in the Module Self-Check Answer Key.
Qi) Which setting is not modified when a user is roaming between sites with a device1!
(Source: Implementing Device Mobility)
A) region
B) directorv number
O location
D) SRST reference
021 Which three of the following are device mobility-related settings in Device Mobility?
(Choose three.) (Source: Implementing Device Mobility)
A) Region
B) SRST Reference
C) AAR Calling Search Space
D) Device Mobility Calling Search Space
F.) Media Resource Group List
F) Location
Ci) AAR Group
03) Which two statements are correct about (he relationship between Device Mobility
configuration elements? (Choose two.) (Source: Implementing Device Mobility'.
A) Device Mobility Infos refer to one or more device pools.
B) Device pools refer to one or more physical locations.
C) Device pools can refer to one device mobility group,
D) Device pools can refer to one Device Mobility Info.
E| Physical locations refer to device mobility groups.
Q4) Which statement is not correct about Device Mobility operation? (Source:
Implementing Device Mobility)
A) A dev ice pool is selected basedon the IP address of the device.
B) If the selected device pool is the home device pool, no changes are made.
C) If the selected device pool is in a different device mobility group than the home
dev ice pool, the device-mobility-related settings of the roaming device poolare
applied.
D) If the selected device pool is in a different physical location than the home
dev ice pool, the roaming-sensitive settingsof the roaming device pool are
applied.
05) Which statement is not correct about the interaction of Device Mobility and globalized
call routing? (Source: Implementing Device Mobility)
A) The user of a roaming phone can use the home dial rules.
B) Fhe user of a roamingphone can use the homedial rules, but then the home
gateway is used all of the time.
C) The user of a roaming phone can use the roaming gateway.
D) The same device mobility group can be used at all devicepools.
©20'0 ico Systems. Inc Implementation of Features and Applications for MultisiteDeployments
4-70 Implementing Cisco Unified Communicalions Manager, Part 2 (CIPT2) v8.0 ©2010 Cisco Systems, Inc.
Mo lule Summary
This lopic summarizes the kev points that were discussed in this module.
Module Summary
Rel frences
For additional infonnation. refer to these resources:
• Cisco Systems. Inc. Cisco Unified Communications System 8.x SRND, April 2010.
hup: www.cisco.conVen/I IS/docvYoiee ip comm/cucm/srnd/Kx/iicSx.htmi
• Cisco Systems. Inc. Cisco Unified Communications Manager Administration Guide
Release 8.0(1). February 2010.
!itlp:.:\\\\w.ei^co.conv'cn/US''docs;voiceJp_comm/cucm/admin/K_0 I.ecmcfg'bccn-SOl-
em.html
• Cisco Systems. Inc. Cisco Unified Communications Manager Features and Services Guide.
Release'8.0{i). March 2010.
imp: www.cisL'o.coin't'iiT^.'docs/voice ip comm/eucm/admin/8 0 l/ccmfeat/fsg^-801-
cm.html
;isco Systems. Inc Implementing Features and Applications for Multisite Deployments 4-69
Summary
This topic summarizes the key points that were discussed in this lesson.
Summary
References
For additional information, refer to these resources:
• Cisco Systems. Inc. Cisco Unified Communications Svstem Release 8.x SRND. San Jose,
California. April 2010.
http:/.'www. cisco.com/en/lJ S/docs/voiee_ip_comm/eucm/srnd/8x/ue8.\srnd.pdf.
• Cisco Systems. Inc. Cisco Unified Communications Manager Administration Guide
Release 8.0(2). San Jose, California, March 2010.
http:.'/v\ww.cisco.com''en/US/docs/voice ip comni/cuem/admin/8 0 2/ccmcfg/bccm.pdf.
• Cisco Systems. Inc Cisco Unified Communications Manager Features andServices Guide
Release 8.0(2). San Jose, California, March 2010.
http://vvw\\xisc().com/en/US/docs/voice__ip_comm/cucni/admin/8J)_2/cciiifeal/fsgd.pdf'.
4-68 Implementing Cisco Unified Communications Manager, Part2 (CIPT2) v8.0 ) 2010 Cisco Systems, Inc
Step 7b: Subscribe Phone to Cisco Extension Mobility Phone
Senvice
The last step of Cisco Extension Mobility configuration is to subscribe the IP phoneto the
Cisco Extension Mobilitv phone service.
1 Logon 1 LcbjI'
FM Lngfm ' Loaon'
FM logon / '-v^fflt
-Sulrairlbed Scrvic
1 Choose the Cisco
Exlension Mobility
-| rvcxl I Close
phone service by
using the name
assigned in Step 3 3. Click Subscribe.
Then, click Next Then, click Save
The process ofsubscribing the IP phone tothe Cisco Extension Mobility service isthe same as
the process that was explained in Step 5. in which the device profile was subscribed tothe
Cisco Extension Mobility service. Inthe Phone Configuration window, use the related link
Subscribe/Unsubscribe Services to open the Subscribed Cisco IP Phone Services window and
subscribe to the sen ice.
) 2010 Cisco Systems, Inc Implementation ofFeatures and Applications for Multisite Deployments 4-67
Step 7a: Configure Phones for Cisco Extension Mobility
Finally, the phone must be enabled for Cisco Extension Mobility and subscribed to the Cisco
Extension Mobility phone service. The figure shows the first part—enabling Cisco Extension
Mobilitv on a phone.
kJ Oct* rt I.
B*.
4-66 Implementing Cisco Unified Communications Manager, Part2 (CIPT2) v8.0 >2010 Cisco Systems, Inc.
Step 6: Associate Users with Device Profiles
This step describes how to associate users withdevice profiles.
1 1
Usei ID and PINaie
l™" —1
] used for Cisco
~< ===== J Extension Mobility login
1
-fr.--.o-* «j-t*f
Associate user
-
/ device profiles,
optionally, set
VA default profile
(if moielhan
one controlled
A
profile exists)
Inthe End User Configuration window (which you can access from Cisco linified
Communications Manager Administration by choosing User Management > End User), choose
the device profile orprofiles that you want to associate with the user in the list of Available
Profiles. Click the down arrow to add them lo the list of Controlled Profiles.
Extension Mobility
- Subscribed Services
phone service by
using the name
assigned in Step 3. 3. Click Subscribe.
Then, cick Next. Then, click Save,
Caution If the device profile is not subscribed to the Cisco Extension Mobility service, users do not
have access to Cisco Extension Mobility phone service after they log in and their device
profile has been applied. As a result, users can no longer log out of Cisco Extension Mobility
at the phone. Therefore, make sure that you do not forget to subscribe the phones (see Step
7b)—and the device profiles—that you use for Cisco Extension Mobility to the Cisco
Extension Mobility phone service.
mm
4-64 ImplementingCisco Unified Communications Manager, Part 2 (CIPT2)vfl.O >2010 Cisco Systems, Inc.
Step 5a: Create Device Profiles
The next step is the configuration of device profiles.
5 modeiand
protoccrf.
© 2010 Cisco Systems. Inc Implementation of Features and Applications for Multisite Deployments 4-63
Step 4: Create Default Device Profiles
If multiple phone models are used for Cisco Extension Mobility, default device profiles should
be enabled.
Note The available configurationoptions depend on the chosen phone model and protocol. The
default device profiledoes not include phone button configuration (for example, lines or
features buttons) but does include the phone button template.
«M
4-62 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 52010 Cisco Systems, Inc.
Step 3: Add the Cisco Extension Mobility Phone Service
fhe next step is to add the Cisco I'xlension Mobility phone service.
<l
Extension Mcbilil. ..ckju Uicif 5trv.c
Mobility service URL.
Semce UHL htrp ''101 i.i.eoeoien OPC'EWAnpSt vlet'devic c-tDI
^?rMrj Catega
Setvice URL
htlp /:Server_IP_Aa<t-ess S0SO'emBpp/EMAppServlePdevice=#OEVICENAMEB
Cisco fxtension Mobility is implemented as a phone service, fherefore,you must add this
service to the available phoneservices in Cisco Unified Communications Manager. To add the
Cisco Extension Mobility phone service, in Cisco Unified Communications Manager
Administration, choose Device> DeviceSettings > Phone Services. Configure the Cisco
Fxtension Mobility service with a service name anddescription, andthenenterthe service
URI.:http:'.V/'rt'r IP ,-W(//-csj:8080.'cinapn/l:MAppScrv!cl?dc\icc=rfI)l'.VICFNAMI;.S.
Note The service URL is case-sensitive and can be found in the Cisco Unified Communications
Manager Help pages.
© 2010 Cisco Systems, Inc Implementation ofFeatures andApplications for Multisite Deployments 4-61
Step 2: Set Cisco Extension Mobility Service Parameters
The Cisco Extension Mobility service has several configurable service parameters.
Cisco Fxtension Mobility can be configured with the following service parameters.
If the Enforce Intra-cluster Maximum Login Time parameter is set to True, the user is
automatically logged out after the Intra-cluster Maximum Login Time expires. The Intra-cluster
Multiple Login Behavior parameter specifies how to process users who log into a device but are
still logged in at another device. There are three options: Login can be denied, login can be
allowed, or the user can be logged out automatically from a phone on which the user logged in
earlier and did not log out.
Alphanumeric User ID can be enabled or disabled, and the last logged in username can be
remembered (and presented as a default on the next login) by setting the Remember the Last
User Logged In parameter to True. Call lists can be preserved or cleared at logout, depending
on the setting of the Clear Call Logs on Intra-Cluster EM service parameter.
Note All of these parameters are clusterwide service parameters of the Cisco Extension Mobility
service and can be accessed from Cisco Unified Communications Manager Administration mm
4-60 Implemenling Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 >2010 Cisco Systems, Inc.
Step 1: Activate the Cisco Extension Mobility Service
fhe first step is to activate the Cisco Extension Mobility feature service.
Se • do
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Cisco CalWanswer Ace rated
ActivateCisco Extension
CISCO Tttp Activated
Mobility service.
Cisco Messaging IrtprTicp Oe actuated
CKfo unrfifd Mobile Vane fides? Ser.-c Attested
Cisco ]P Vo<e. Media Streamy Aoo Ac! vsted
lo enable Cisco Extension Mobility, you must activate the Cisco Extension Mobility feature
service from Cisco Unified Serviceability. To do so, click Tools > Service Activation.
Note Starting with Cisco Unified Communications Manager Version 6.0, Cisco Extension Mobility
is considered a User Facing Feature and can be activated on any server in a Cisco Unified
Communications Manager cluster, to provide a redundant Cisco Extension Mobility
environment
*>
© 2010 Cisco Systems. Inc. Implementation of Features and Applications for Multisite Deployments 4-59
Cisco Extension Mobility Configuration
This topic describes how to configure Cisco Extension Mobility.
The figure lists the steps that are required to configure Cisco Extension Mobility in Cisco
Unified Communications Manager. The following topics explain these steps in detail.
4-5S Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 12010 Cisco Systems, Inc.
Alternatives for Mismatching Phone Models and CSS
Implementations
To avoid issues with mismatching IP phone models or with calling privileges when the
traditional approach for implementing partitions andCSSs is used, multiple device profiles can
be configured per user.
When different phone model series are used, issues can arise when the settings of the default
device protlle are applied. DilTerenl users might require different settings. This problem can be
solved by creating multiple device profiles peruser. When you configure and associate one
device profile (per phone model) with a username, Cisco Unified Communications Manager
displavs this list ofprofiles after successful login. The user can choose a device profile that
matches the phone model ofthe login device. Ilowever. ifmany users need touse Cisco
E\tension Mobility and many different phone models arcused, this solution does notscale
well.
The same concept can be used as an alternative tothe line/device approach for implementing
CSSs. Aseparate device profile can be created per site and is configured with the appropriate
CSS to allow local gatewa> s to be used for external calls. Again, the user chooses the
corresponding device profile after logging in. and the correct CoS and gateway choice are
applied without depending on a separate line and device CSS. The recommendation, however,
isto use the line/device approach in a multisite environment, because that approach simplifies
the dial plan and scales belter.
Note When using thetraditional CSS approach with only oneCSS applied at theline, use Local
Route Groups to prevent gateway-selectionproblems.
- For proper gateway selection, use Local Route Groups if the CSS is
applied at the phone line.
• AAR CSS is configurable only at the device and is never updated by Cisco
Extension Mobility, so local gateway can be used for AAR calls.
Cisco Extension Mobility does not modify the device CSS or the automated alternate routing
(AAR) CSS (both of which are configured at the device level). Cisco Extension Mobility does
replace the line CSS or CSSs that are configured at the phone with the line CSS or CSSs that
are configured at the device profile of the logged-in user.
Thus, in an implementation that uses the line/device approach, the following applies:
• The line CSS of the login device is updated with the line CSS of the user. This update is
used to enforce the same class of service (CoS) settings for the user, independent of the
physical device to which the user is logged in.
• The device CSS of the login device is not updated, and the same gateways (those gateways
that were initially configured at the phone before the user logged in) are used for external
route patterns. Because the phone did not physically move, the same local gateways should
be used for PSTN calls, even when a different user is currently logged into the device.
If the traditional approach is used to implement partitions and CSS, the following applies:
• If only device CSSs are used, the CSS is not updated, and no user-specific privileges can be
applied. The user inherits the privileges that are configured at the device that is used for
logging in.
• If only line CSSs are used, the line CSS that is configured at the device profile of the user
replaces the line CSS of the login device. In a multisiteenvironment, this configuration can
cause problems in terms of gateway choice because the same gateway is always used for
external calls. To avoid gateway selection problems in such an environment, you should
use Local Route Groups.
4-56 ImplementingCisco Unified Communications Manager, Part 2 (CIPT2)v8.0 © 20t0 Cisco Systems, Inc.
2. The phone button template and the softkey template of the default device profile are
applied to the Cisco Unified IP Phone 7905.
3. The user has access to the phone services that are configured in the Cisco Unified IP Phone
7905 default de\ ice profile.
mm
) 2010 Cisco Systems, Inc Implementation ofFeaturesand Applications forMullisite Deployments 4-55
How Cisco Extension Mobility Handles Phone Model
Differences
This subtopic describes the phone configuration process when Cisco Extension Mobility isused
with different phone models.
After successful authentication, if the phone model series of the device protlle does not match
the phone model series of the used phone, the following happens:
1. Device-dependent parameters, such as phone button template and softkey template, from
the default device profile are applied lo the phone.
2. Then the system copies all device-independent configuration settings (user hold audio
source, user locale, speed dials, and line configuration, except for the parameters that are
specified under I inc Setting for This Device) from the device profile to the login device.
3. Next, the applicable device-dependent parameters of the device profile of the user are
applied. These parameters include buttons(such as line and feature buttons) that are based
on the phone button template that has been applied from the default device profile.
4. Finally, if supported on the login device, phone service subscriptions from the device
profile of the user are applied to the phone.
5. If the device profile of the user does not have phone services that arc configured, the
system uses the phone services that are configured in the default device profile of the login
device.
For example, the following events occur when a user who has a device profile for a Cisco
Unified IP Phone 7960 logs into a Cisco Unified IP Phone 7905:
I. The personal user hold audio source, user locale, speed dials (if supported by the phone
button template that is configured in the Cisco Unified IP Phone 7905 default device
profile), and directory number configuration of the user are applied to the Cisco Unified IP
Phone 7905.
4-54 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) vB.O >2010 Cisco Systems, Inc.
Default Device Profile and Feature Safe
This subtopic describes the feature safe functionality of Cisco Extension Mobility.
Ihe default deuce profile is applied only ifa users device profile and the phone on which the
user tries to log in are of a different phone model scries (for example, Cisco Unified IP Phone
Scries 794x. 796x. or 797x).
When the phone model scries of the physical phone and the user device profile are the same,
the feature safe function allows different phone models lo be used for user device profiles and
physical phone models.
for example, a user with an associated device profile for a Cisco Unified IP Phone 7940 phone
can log into a Cisco Unified IP Phone 7945 phone without having the default device profile
applied.
No administrative tasks are required to enable feature safe. Feature safe is independent of the
used signaling protocol (SIP or SCCP).
© 2010 Cisco Systems. Inc Implementation of Features and Applications for Multisite Deployments
Issues in Environments with Different Phone Models
This subtopic describes issues with mixed IP phone environments.
When different IP phone models are used in a Cisco Unified Communications Manager cluster
for which Cisco Extension Mobility is enabled, an end user may log into an IP phone that is of
a different model series than the one that is configured in the device profile of the user.
Different phones support different features. Therefore, when a user logs into a phone that
supports more features than are supported by the model that is associated with the user, the
default device profile is used to apply the parameters that the target phone supports but that are
not included in the device profile of the user. The default device profile includes phone
configuration parameters such as phone button templates, softkey templates, phone services,
and other phone configuration settings. However, the profile does not include button
configuration (including line buttons).
4-52 ImplementingCisco Unified Communications Manager, Part 2 (CIPT2)v8.0 >2010 Cisco Systems, Inc.
5. The IP phone is reset and loads the updated configuration.
Now. the phone can be used as it would be used in the home location. Directory' numbers,
speed dials. MWI. and so on. are all correct, regardless of the location and the IP phone that is
used.
Users can log out of Cisco Extension Mobility by pressing the Services button and choosing
Logout in the Cisco Extension Mobility service. If users do not logout themselves, the system
automatically logs them out after the expiration of the maximum login lime (if the appropriate
sen ice parameter has been configured accordingly).
The user is also automatically logged out of a phone when the user logs into another phone and
when Cisco Unified Communications Manager is configured for auto-logout on multiple
logins. Another option is that the next user of the phone logs out a previous user so that the new
user can log in and have the phone that is updated with the settings of that new user. After
logout. Cisco Unified Communications Manager reconfigures the phone either with the
standard configuration of the IP phone or by using another device profile (as specified in the
Phone Configuration window).
© 2010 Cisco Systems. Inc. Implementation ofFeatures andApplications for Multisite Deployments 4-51
Cisco Extension Mot ility Operation
This topic describes how Cisco I xtension Mobility works, how phone model mismatches are
processed, and how calling scare l spaces (CSSs) and partitions areupdated when Cisco
Extension Mobilitv is used.
When a userwants to log into a phone, the following sequence of events occurs:
1. The userpresses the Services button on the phone andchooses the CiscoExtension
Mobility service from the list of phone services that are available onthe phone.
2. The Cisco Extension Mobility service requires the user to log inby using a user ID and
PIN. The user enters the required data.
3. Ifthe entered user ID and PIN are correct, Cisco Extension Mobility chooses the device
profile that is associated with the user.
m
Note If a user isassociated with more than onedevice profile, all associated profiles are
displayed andthe user must choose thedesired profile. Assigning multiple profiles toa user m
meansthatthe useris provided a separate device profile per site. This approach is common
when thetraditional approach is used toimplement CSSs. Cisco Extension Mobility updates
mm
onlythe lineconfiguration (including the line CSS), not the device CSS. To allowthe choice
ofa local gateway, a different (line) CSS must be applied persite. In such a scenario, the
userchoosesa site-specific device profile thatdiffers from the device profile thatis usedat
other sites in theline CSS. The line CSS ofsuch site-specific profiles gives accessto route
patterns that route public switched telephone network (PSTN) calls tothe appropriate (local)
gateway
4. Cisco Unified Communications Manager updates the phone configuration with the settings
ofthe chosen device profile. User-specific device-level parameters, lines, and other phone
buttons areupdated with user-specific settings.
4-50 Implementing Cisco Unified Communications Manager, Part 2 (C1PT2) v8.0 ) 2010 Cisco Systems, Inc
Relationship of Cisco Extension Mobility Configuration
Elements
The figure shows how the Cisco Extension Mobility configuration elements relate toeacl
other.
'••'tm&k —
Device
Profile: A
Cisco Extension
MobilityPhone
Service
Oevice
Stilt N— Profile: B
\
\
Device
Profile: C
As the figure shows, an end user isassociated with one ormore device profiles. For each
possible IP phone model and protocol (SCCP and SIP), adefault device profile can be
configured. Because Cisco Extension Mobility is implemented as aCisco IP Phone Seniee. all
phones that should support Cisco Extension Mobility must be subscribed to the Cisco
Extension Mobility phone service, loallow a user to log into the phone. In addition, each
mm
device profile must be subscribed to the Cisco Extension Mobility phone service; this
subscription isrequired to allow a user to logoutof a phone.
Note Cisco Unified Communications Manager automatically creates a default device profile for a
specific phone model and protocol, as soon as Cisco Extension Mobility is enabled on any
phone configuration page for this phone model.
4-48 Implementing Cisco Unified Communications Manager, Pari 2(CIPT2) v8.0 ©2010 Cisco Systems, Inc
Cisco Extension Mobility Configuration Elements
Thistopic describes the configuration elements thatCisco Extension Mobility uses.
The figure lists the configuration elements that are related toCisco Extension Mobility and
describes theirfunction. Theconfiguration elements that are introduced with Cisco Extension
Mobility are the device profile and the default device profile.
The device profile isconfigured with all the user-specific settings that are found althe device
level ofan IP phone (user MOH audio source, phone button templates, softkey templates, user
locales. DND and privacy settings, and phone service subscriptions) and all phone bulto.s
(lines, speed dials, and so on). One ormore device profiles are applied toan end user. ir> the
End User Configuration window.
The default dev ice profile stores default device configuration parameters thatCisco Extension
Mobilitv applies when there isa mismatch ofthe phone model series on which the user logs in
and the phone model series that is eonligured in the device profile ofthe user. The default
device profile exists once per phone model type and per protocol (Session Initiation Protocol
[SIP] and Skinny Client Control Protocol [SCCP]). All ofthe parameters that cannot be applied
from the device profile of the user are taken from the default device profile.
Eor example, a user is associated with adevice profile for aCisco Unified IP Phone 7945 that
runs SCCP. Ifthis user logs in toa Cisco Unified IP Phone 7965 that runs SIP. some features
(configuration parameters) that exist on the target phone are not configurable on the Cisco
Unified IP Phone 7945 dev ice profile. Inthis case, theconfiguration parameters that arc
unavailable onthe device profile of the user are taken from the default device profile ofthe
Cisco Unified IP Phone 7945 SCCP.
Ifadevice profile includes more parameters than the target phone supports, the additional
settings are ignored when the target phone with the user-specific settings is reconfigured.
Cisco Unified
Communications
Remote
Manager
Gateway
Logout Login
H[
UMTUKda
Iter yOHAufe Sara and PIN UwrMOH AMID South
Ftcru tUscr, Timvlm ftm* fcltwt'r«iwl*»
SetmTtmme
NJ
Unicss \limeas
As shown in the figure, the user-specific parameters (that is, some device-level parameters and
all phone button settings, including line configuration) are configured in device profiles. Based
on the user ID that is enteredduring login,Cisco Unilied Communications Manager can apply
the personaldevice profileof the user and can reconfigure the phone with the configuration
profile of the user who logs in.
With Cisco Extension Mobility, CiscoUnified Communications Manager is aware of the end
userof a device andapplies the appropriate user-specific configuration, according lo a device
profile that is associated with the logged-in user.
4-46 Implementing CiscoUnified Communications Manager, Part 2 (CIPT2) vfi.O >2010 Cisco Systems, Inc.
Cisco Extension Mobility: Dynamic Phone Configuration
Parameters
Ihis subtopic describes the parameters that arc updated when a user logs in to a phone wnen
using Cisco Kxtension Mobility,
There arc two types of configuration parameters that aredynamically configured when Cisco
Extension Mobility is used:
• User-specific de\ice-le\el parameters:
— fhese user-specific phone configuration parameters include user Music on Hold
(MOH) audio source, phone button templates, softkey templates, user locales. Do
Not Disturb (DND) andprivacy settings, and phone service subscriptions. All these
parameters are configured at the device level of an II' phone.
• Configuration of phone buttons(including lines):
— Cisco Extension Mobility updates all phone buttons -not only the button types that
are specified inthe phone button template but also the complete configuration of the
phone buttons. This update includes all configured lines, with all the line
configuration settings, speed dials, service UKUs. Call Park buttons, and any other
buttons that are configured in the device profile lhal is to be applied.
©2010 Cisco Systems Inc Implementation ofFeatures and Applications for Mullisite Deployments 4-45
Cisco Extension Mobility
Characteristics (Cont.)
If a user logs in with a user ID that is still logged in at another device, one of the following
options can be configured:
• Allow multiple logins: When this method is configured, the user profile is applied to the
phone on which the user is logging in. The same configuration remains active at the device
on which the user logged in before. The line number or numbers become shared lines
because they are active on multiple devices.
• Deny login: Whenthis optionis configured, the user receivesan error message. Login is
successful only after the user logs out of the other device on which the user logged in
before.
On a phone that is configured for Cisco Extension Mobility, anotherdevice profile (a logout
device profile) canbe applied, or theparameters that are configured on the phone are applied.
The logoutcan be triggered by the user or enforced by the system after expiration of a
maximum login time.
4-44 Implementing CiscoUnified Communications Manager, Part2 (CIPT2) v8.0 >2010 Cisco Systems, Inc.
Cisco Extension Mobility Overview
This topic describes the keycharacteristics and features of Cisco Extension Mobility.
- Allows users to log into any phone and have their user-specific
phone configurations applied to the phone
• Makes users reachable at their personal extensions,
regardless of their locations and the physical phones they use
« Is implemented as a phone service; works within a Cisco
Unified Communications Manager cluster
- Stores user-specific phone configuration in device profiles
Cisco Extension Mobility allows users to log in to any phone and have theirindividual, user-
specific phone configuration that isapplied tothat phone. Thus, users can be reached at their
personal directory number, regardless of their location or the physical phone that they are
using, Cisco Extension Mobility is implemented asa phone service and works within a Cisco
Unilied Communications Manager cluster.
The user-specific configuration is stored in device profiles. After successful login, the phone is
reconfigured with user-specific parameters; other (device-specific) parameters remain the same.
Ifa user is associated with multiple device profiles, the user must choose which device profile
to use.
Speed dials are assigned lo physical devices. Spaed dials are assigned to device profiles.
Services are assigned tophysicai devices. Services are assigned to device prolles.
casng privileges are delned for phytic*! ^ »*'*«••'**>* "™ mar9a *lln9
devices andlocations tttttoa.* Wevfce-based) Bfldptiyscal device
oevicej analocaons. umg.s(locatiCKKef*«d).
4-42 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) vB.O ) 2010 Cisco Systems, Inc.
Issues of Roaming Users
Using guest phones at remote sites leads to several issues.
The figure lists the most common issues that arise when users use any available guest phone at
sitesto which they have traveled. These issues include wrong extension numbers andcalling
privileges, other speed-dial configuration and phone-sen.'ice assignments, and no MWI status
for the actual number of the user.
For correct settings, the user requires Cisco Unified Communications Manager to reconfigure
the used phone with user-specific configuration instead ofhaving device-specific settings that
are applied to the phone.
© 2010 Cisco Systems, Inc Implementation otFeatures and Applications for Multisite Deployments 4-41
Issues with Users Roaming Between Sites
This topic describes the issues that can occur when users temporarily change their workplaces
and roam between sites.
Roaming Users
Cisco Unified
Communications
Remote
Manager
Gateway
Roaming User
- - - - . : - _ . u—
Whenusers roam between sites and do not have their phone with them (for example, via Cisco
IP Communicator), they might want to use any available phone at the site to which they have
traveled.
4-40 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 12010 Cisco Systems. Inc.
Lesson 2
Objectives
Upon completing this lesson, you will be able to describe how Cisco Extension Mobility works
and how it is implemented. This ability includes being able to meet these objectives:
Identity the issues when users roam between sites
Describe the Cisco Extension Mobility feature
Describe the Cisco Extension Mobility configuration elements and their interaction
Describe Cisco Extension Mobility operation
Implement Cisco Extension Mobility
m
4-38 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 © 2010 Cisco Systems, Inc.
References
I'or additional infonnation. refer to these resources:
• Cisco Systems. Inc. Cisco Unified Communications System 8.xSRND. April 2010.
hup: '.www.cisco.coni en US/doesAoice_ip comm/aicm/srn<l/8,\/uc8x.html
• Cisco Systems. Inc. Cisco Unified Communications Manager Administration Guide
Release 8.0(1). February 2010.
hup: 'www.cisco.com en US/docs/\oice ipcoin nv'cu em/admin/8 0 l/ccmcfg/beem-XOI
cm.html
)2010 Cisco Systems. Inc Implementation of Features and Applications for Multisite Deployments
Summary
'ITiis topic summarizes the key points that were discussed in this lesson.
Summary
Summary (Cont.)
4-36 Implementing Cisco Unified Communications Manager, Part2 (CIPT2) v8.0 52010 Cisco Systems, Inc.
Step 5b; Set the Device Mobility Mode for Individual Phones
The fig jrc shows how to set the Device Mobility mode for each phone.
H
Set the phone device mobilitymode: .
In the I'honc Configuration window, you enable ordisable Device Mobility for each phoie by
cither setting Device Mobility Mode to On orOff or leaving the default value as Default. If
Device Mobilitv Mode is setto Default, the Device Mobility mode that is setat theCisco
CallManager serviceparameteris used.
The figure also shows the configuration ofthe overlapping parameters (these arc parameters
that can be configured at the phone and at the device pool). The overlapping parameters for
roaming-sensitive settings are Media Resource Group List, Location, and Network Locale. The
overlapping parameters for the Device Mobility-related settings are Calling Search Space
(called Device Mobility Calling Search Space atthe device pool), AAR Group, and AAR
Calling Search Space. Overlapping parameters that are configured at the phone have higner
prioritv than settings at the home device pool, and lower priority than settings at the roaming
device pool.
£( '.'.;-.i,;i[i;: * off u c«
Qt- -s!•;!f.. 1SJ* " n„ ,«|
^u^^, A
y\
Set the default Device Mobility
mode for all phones.
Device Mobility is turned off by default and is configurable for eachphone. To set the default
for the Device Mobility mode (ifit isnot setdifferently atthe phone), choose System >
Service Parameter. ChooseCisco CallManager, and in the Clusterwide Parameters
(Device—Phone) section, set Device Mobility Modeto On or Off (OtTis the default).
4-34 Implementing Cisco Unified Communications Manager. Part2 (CIPT2) v8.D >2010 Cisco Systems. Inc.
Step 4: Configure Device Mobility Infos
The figure shows how to configure device mobility infos.
Io configure device mobility infos, choose System > Device Mobility >Device Mobility-
Info. They are configured with aname, a subnet, and a subnet mask. Then they are associated
with one or more device pools.
Cisco Unified
Communications
Manager Ad ministration:
System > Device Pool
ConGguteroaming-
sensitive sellings mat <
be applied lo Ihe ptione
configuration The Local
Route QOLp ib also a
roaming -sens itiveset! ing'
You configure a device pool with aname and a Cisco Unified Communications Manager
group. The configuration includes roaming-sensitive settings and Device Mobility-related
settings. (You configure the Device Mobility-related settings in the Device Mobility-Related
Information section.) You configure both the physical location and the device mobility group in
the Roaming-Sensitive Settings section. You use both of those configurations todecide which
settings to apply to a phone: nosettings, the roaming-sensitive settings only, orthe roaming-
sensitive settings and the Device Mobility-related settings. Thephysical location and the device
mobility group themselves arenotapplied to the configuration of a phone, butareused to
control which settings to apply.
4-32 Implementing Cisco Unified Communicalions Manager, Part 2 (CIPT2) vS.O >2010 Cisco Systems, Inc.
Steps 1 and 2: Configure Physical Locations and Device
Mobility Groups
fhe figure shows the configuration of physical locations and device mobility groups.
••- • •
To configure physical locations, choose System > Physical Location, for each physical
location, you configure a name and a description. To configure device mobility groups, choose
System > Device Mobility > Device Mobility Group. For each device mobility group, you
configure a name and a description.
Note Device mobility groups are not necessary when there is no need to change the device level
CSS, AAR CSS, and AARgroup This principle applies also when local route groups are
used in an environment where all sites share the same dial rules or in an environment where
globalized call routing is implemented
© 2010 Cisco Systems, Inc Implementation of Features and Applications for Multisite Deployments
Device Mobility Configuration
This topic describes how to configure Device Mobility.
"Die figure lists the required steps for implementing Device Mobility. As discussed in the
previous topics, device mobility groups are not required when you implement Device Mobility
in an environment where globalized call routing is used.
4-30 Implementing CiscoUnified Communicalions Manager, Part 2 (CIPT2) v8.0 >2010 Cisco Systems. Inc.
Example: Globalized Call Routing
The figure shows an example of Device Mobility in an environment where globalized call
routing is implemented. Also, gateway selection is performed by the local route group feature.
Theexample in the figure is based on the previous scenario: IIt) is in Europe. BRis in the
United States. A BR user will roam to Europe.
However, in this example, globalized call routing has been implemented. Therefore, the (line)
CSS of BR phones provides access to translation patterns thatconvert localized call ingress at
the phone (NANP formal) to global E.I64 formal. EU phones have access to translation
m
patterns that cotwert HI] input to global K.164 fonnat.
A single PSTN route pattern (\(!) is configured: it is in a partition that is accessible by all
translation patterns.
When a BR user roamsto the IIQ. the lineCSS is not modified; no device CSS is configured at
the phone or at the device pool. Thedevice mobility groups arc alsonotset (or areset
differently).
As a result, there is effectively no change in matching the translation patterns: The BRuserstill
uses NANPdial rules (like at home), fhe numberis converted to international format by-
translation patterns and matches the (only) PSTN roule pattern. The route pattern refers to a
route listthat is configured to use thedefault local route group. The default local route group is
taken from the roaming device pool. Therefore, if the phone is physically located in the BR
office, the local route group is BR; if the phone is roaming to the HQ site, the local route group
is HQ. As a result, the local gateway is always used for a PSTN call.
IfTEHO was configured, there would be aTEHO route pattern in E.I64 format with a leading
+sign. The TEHO pattern would refer toa site-specific route list in order toselect the correct
gateway for PSTN egress. The backup gateway would then again be selected by the local route
group feature.
) 2010 Cisco Systems. Inc Implementation of Featuresand Applications for Multisite Deployments 4-29
Example: No Globalized Call Routing—Same Device Mobility
Group
The figure shows an example of Device Mobility with identical device mobilitv groups in an
environment where globalized call routing is not implemented. Also, gateway selection is
performed by the device CSS ofthe IP phone.
Device Pool HQ
Physical
Location. HQ
Device Mobility
Group; World
Device Mobility
CSS' HQ
This example is identical to the previous example with one exception: This time the device
mobility group ofthe home and the roaming device pool are the same.
When a BR user roams to the HQ, the device CSS ofthe phone is updated with the device CSS
of the roaming device pool. In the example, CSS BR is changed to HQ. As aconsequence the
phone has access to the HQ partition that includes PSTN route patterns in EU dialing format
Therefore, the roaming user has to follow EU dial rules. Calls to 9.@ are not possible anymore
However, this configuration allows the BR user to use the HQ gateway when roaming to the
HQ.
•fe
4-28
Implemenling Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 ^2010 Cisco Systems, Inc.
Example: No Globalized Call Routing—Different Device
Mobility Group
The figure shows an example of Device Mobility with dilTerenl device mobility groups in an
em ironment where globalized call routing is not implemented. Also, gateway selection is
performed b\ the device CSS ofthe IP phone.
Device Pool BR
Route Pattern Physical
9® Location1 Lit*
Partition Branch Device Mobility
Group. BR
(Device Mobility
Route List CSS. BR)
Roule Group. BR-GW
HQ: EU Numbering
Plan
UR. NANP
In the example, there are two sites. The main site ("I IQ" in the figure) is in Europe, the branch
site ("BR"' in the figure) is in the United States. Separate route patterns (representing the
different dial rules) are configured in different partitions. The CSS of HQ phones provides
access to the HQ gateway, the CSS ofBR phones provides access to the RR gateway.
Device Mobilitv is configured with different device mobility groups. This configuration allows
BR users who arc roaming with their phones to the HQ to use the home d.al rules. Ihe device
CSS is not updated by Device Mobility, and therefore, the CSS still provides access to the BR
route pattern (9,ff). Howler, as aconsequence, the BR gateway ,s used tor all PS fN calls.
When Device Mobility with globalized call routing is used, there are no changes in the
roaming-sensitive settings. Their application always makes sense when roaming between sites
Iliey have no influence on the gateway selection and the dial rules that auser has to follow.
The dial plan-related part of Device Mobility, however, changes substantially with globalized
call routing. It allows aroaming user to follow the home dial rules for external calls and
nevertheless utilize the local gateway ofthe roaming site.
This situation is possible because globalization of localized call ingress at the phone occurs
1his function is provided by the line CSS of the phone. It provides access to phone-specific
translation patterns that normalize the localized input ofthe user to global format. The device
CSS that was used for gateway selection is obsolete, because gateway selection is now
performed by the local route group feature.
The AAR CSS and AAR group that are configured at the device level can be the same for all
phones as long as the AAR number is always in global format. (You can ensure that it is always
in global format by configuring either the external phone number mask orthe AAR
transformation mask to E. 164 format.) In this case, no different AAR groups arc required
because there is no need for different prefixes that are based on the location of the two phones
Further, there is no need for different AAR CSS, because the gateway selection is not based on
d.fferent route lists (referenced from different route patterns in different partitions) Instead it
is based on the local route group that was configured at the device pool of the calling phone.
In summary, when using globalized call routing is used, Device Mobility allows users lo use
local gateways at roaming sites for PSTN access (or for backup when TEHO is configured)
while utilizing their home dial rules. There is no need to apply different device CSS AAR CSS
and AAR groups, and hence, device mobility groups are no longer required
In summarv. unless TEHO is used, the implementation ofDevice Mobility without globalized
call routing leads to this situation: Either the home gateway has to be used (when allowing the
user touse the home dial rules) orthe user is forced touse the dial rules ofthe roaming site (in
orderto use the local gateway of theroaming site).
m*
Local route groups have been introduced with Cisco Unified Communications Manager
Version 7. When local roule groups—and globalized call routing, which utilizes local route
groups—is not used or supported. Device Mobility is typically implemented:
• Roaming-sensitive settings are always updated when the device roams between different
physical locations. These settings are location, region, SRST reference, MRGL, and other
parameters that do not affect the selection of the PSI'N gateway or the local rules.
• Device Mobility-related settings can be appliedin addition to the roaming-sensitive
settings (which meansthat a phone has to roam between differentphysical locations). The
Device Mobility-related settings are device CSS, AAR CSS, and AAR group. The
configuration of the device mobility group shoulddetermine your decision about whether
to apph the Device Mobility-related settings.
— If the device roams between different device mobility groups, the Device Mobility-
related settings arenot updated with the values that were configured at the roaming
de\ ice pool. This configuration hasthe advantage thatusers do not have to adaptto
different dial rulesbetween homeand roaminglocation (if they exist).The
disadvantage is that all PS'fN calls will use the homegateway that can leadto
suboptima! routing.
— Ifdie device roams within the same device mobility group, the Device Mobility-
related settings areupdated with the values of the roaming device pool. This
configuration has the advantage that all PSTN calls will use the local (roaming)
gateway, which is typically desired for roamingusers. However, the users will have
to use to the local dial rules.
4-24 Implementing CiscoUnified Communications Manager, Part 2 (CIPT2) v8.0 >2010 Cisco Systems, Inc.
For the next three scenarios, the assumption is that the home device pool and the roaming
device pool are assigned to different DMGs. As a result, the Device Mobility-related settings
are not applied. Therefore, calls that are placed from the roaming device are routed in the same
way as when the device is in its home location.
• If the user places a call to a UK PS'fN destination, the call will use the IP WAN to the
London site and break out to the PSTN at the London gateway with a local or national call.
This solution is the optimal one from a toll perspective.
• If the user places a call to a PSTN destination that is close to the roaming location (for
example, a U.S. PSTN number) and TEHO is not configured, the call will use the IP WAN
from the U.S. office to the I .ondon site. It wilt break out to the PSTN at the London
gateway to place an international call back to the United States. From a toll perspeebve,
this is the worst possible solution, because the call first goes from the United States to
London over the IP WAN (consuming bandwidth) and then goes back from London to the
United States via a costly international call.
• If the user places a call to a PSTN destination that is close lo the roaming location (for
example, a U.S. PSTN number) and TEHO is configured, the U.S. gateway is used for a
local or national call. This event is optimal from a loll perspective.
Note In these three examples, the user has to dial PSTN destinations by following the dial rules of
the home location (Great Britain).
In summary, when you allow the Device Mobility-related settings to be applied (by using the
same device mobility group), calls to the home location will use a local PS'fN gateway to place
a long-distance or international call when not implementing TEI10.All othercalls areoptimal.
When the Device Mobility-related settings arc not applied (by using different device mobility
groups) and TEHO is not used, calls to the roaming location will first usethe IP WAN to go
from the roaming location to the home location, fhe calls then use the home gateway to place a
long-distance or international call back to the roaming location. All othercalls areoptimal.
The discussed scenarios assume that globalized call routing and local routegroupsare not used,
fhe impact of globalized call routing and local route groups is discussed inthe next topic.
©2010 Cisco Systems, Inc. Implementation ofFeatures and Applications for Multisite Deployments 4-23
Examples of Call-Routing Paths Based on Device Mobility
Groups and TEHO
The table shows how calls are routed in different Device Mobility scenarios.
Same device mobity group, call lo PSTN Cal use* local PSTN gateway at roaming
destination close id dome location, no TEHO. location for a long-distance PSTN cal.
Same device mobility group, call lo PSTN Cal uses IP WAN to gateway at home location
destination close D home location, TEHO. for a local PSTN call.
Same device mobity group, call to PSTN Cal uses local PSTN gateway at roaming
destinalIon close b roamhg location. location for a local PSTN call.
Different device motility group, call to PSTN Cal uses IP WAN to gateway at dome local on
destination close lo home location. for a local PSTN call.
Different device mobility group, call lo PSTN Cal uses IP WAN to gateway at home location
destination close Id roaming location, no TEHO. tor a king-a stance PSTN caK.
Different 0evice mobility group, call lo PSTN CaB uses local PSTN gateway at roaming
destination close to roammg location, TEHO. location for a local PSTN call.
Calls are routed differently depending on the configuration of DMGs (whether Device
Mobility-relatedsettings are applied or not), the dialed destination, and the use of tail-end hop-
off (TEHO). In some scenarios, calls might take suboptimal paths.
The example in this discussion assumes that a user from London roams to the U.S. office (for
simplicity, it is assumed that there is only one U.S. office). This user uses Cisco IP
Communicator.
Forthe first three scenarios, the assumption is that the home devicepool and the roaming
devicepool are assigned to the same device mobility group.Therefore, Device Mobility applies
Device Mobility-related settings. As a result, PSTN calls that are placed from the roaming
device are treated like PSTN calls of standard U.S. phones.
• If the user places a call to a PSTN destination that is close to the home location (for
example, a UK PSTN number) and TEHO is not configured, the call will use the local
(U.S.) PSTN gateway for an international PSTN call, from a toll perspective, this is a
suboptimal solution, because the IP WAN is nol used as much as it could be used when
implementing TEHO. This factor applies not only to the roaming user, but also to U.S.
users who place calls to PSTN destinations in Great Britain.
• If the user placesthe same call (to a UK PSTN number) and TEHO is configured, the call
will use the IP WAN to the London site and breakout to the PSTN at the London gateway
with a local call. This solution is the optimal one from a toll perspective.
• If the user places a call to a U.S. destination number, the U.S. gateway is used fora local or
national call. This event is optimal from a toll perspective.
Note In all ofthe examples that are shown inthe table, the user has to dial PSTN destinations by
following the U.S. dial rules (North American Numbering Plan [NANP]).
4-22 Implementing Cisco Unified Communicalions Manager, Part 2 (CIPT2) v8.0 © 2010 Cisco Systems, Inc.
The line/device approach is recommended when you are implementing CoS in a multisite
environment. Here is a description of the operation of the line/device approach:
• The line CSS implements CoS configuration by permitting internal destinations (other
phone director; numbers, and access to features such as call part and Meet-Me
conferences). The line CSS also blocks PSTN destinations. Because the line CSS is not
changed by Device Mobility. CoS settingsof the device are kept when the device is
roaming.
• I he de\ice CSS is modified when the device is roaming within the same device mobility
group. In this case, the device CSS that is used at the home location is replaced by a de\ ice
CSS that is applicable for the roaming location. This device CSS will refer to the local
gateway of the roaming site instead of to the gateway that is used at the home location.
If the traditional approach is used (only one CSS. combining CoS and gateway choice), the
device CSS must be used. The reason is that Device Mobility cannot modify the line CSS. and
the line CSS has priority over the device CSS (which can be modified by Device Mobility).
The AAR CSS is configurable only at the device level and therefore is always correctly
replaced when the device roams between physical locations within the same device mobility
group.
Note When using globalizedcall routing and local route groups, there is no need for site-specific
device-level CSS. More information about the interaction of globalized call routing and
Device Mobility is provided in a later topic of this lesson.
>2010 Cisco Systems. Inc Implementation of Features and Applications for Multisite Deployments 4-21
Device Mobility and Calling Search Spaces
This subtopic describes how CSSs are processed when Device Mobility is used.
fhese two CSSs allow the use of the line/device approach for implementing calling privileges
and the choice ofalocal gateway for PSTN calls. With the line/device approach, all possible
PS INroute patterns exist once per location and are configured with asite-specific partition
This partition is included in the device CSS ofthe phones, and therefore itenables the use ofa
local gateway for PSTN calls. To implement class ofservice (CoS), PSTN route patterns that
should not be available to all users (for example, international calls, long-distance calls or all
toll calls) are configured as blocked route patterns. These blocked route patterns arc then
assigned to separate partitions. The line CSS ofaphone now will include Ihe partitions of those
route patterns that should be blocked for this phone. Because the line CSS has priority over the
device CSS. the blocked pattern will take precedence over the routed pattern that is found in a
partition that is listed at the device CSS.
4-20
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) u8.0 ©2010 Cisco Systems, Inc.
Gemian users who roam with their softphoncs tothe United Stales might beconfused when
they obtain their home extensions, but they have to use U.S. dialing rules (access code 9instead
of6. or011 instead of00 for international numbers, and so on). Ifyou want to avoid such
behavior, vou need to suppress the application of Device Mobility-related settings. You
suppress the settings by assigning device pools that are to be used at sites with different dialing
rules into different device mobility groups (and in different physical locations). When a user
now roams with a device from Germany tothe United States, all the roaming-sensitive settings
are applied to use local media resources and Cisco Unified SRST gateways. Also, codecs and
CAC settings are applied correctly, but the Device Mobility-related settings arc not applied.
The result is that the phone will use the PSTN gateway and dial rules ofits home location even
though the user has moved to another site. The user does not have to adapt to the dial rules of
the local site to which the phone was moved.
Note The preceding statements regarding call routing and dial behavior that are based on Device
Mobility-related settings do not apply when globalized call routing is used. Alater topic in
this lesson presents more information about the interaction of globalized call routing and
Device Mobility
Ho10 Cisco Systems. Inc. Implementation of Features and Applications for Multisite Deployments 4-19
Device Mobility Considerations
This subtopic describes key facts that you need to consider when implementing Device
Mobility.
• Roaming-sensitive settings:
- Ensure the use of local media resources and SRST references.
- Ensure correct use of codecs and CAC between sites.
- Shouldalways be applied to roaming devices
• Settings relatedto Device Mobility affect call routing:
- WhatgatewaytouseforPSTNaccessandAAR PSTN calls(device CSS
andAAR CSS),and how to composethe AAR number(AAR group)?
- Changes may result indifferent dialing behavior (forexample, different
PSTN accesscodes, different PSTN numbering plans, and so on).
- Users might getconfused byhaving their home extensions and yet being
required to follow dial rules of roaming site.
- tfthisis notdesired, suppress application of settings related to Device
Mobility by assigningdifferent device mobility groups.
- Not applying settings related to Device Mobility might lead to suboptimal
call-routng paths.
Roaming-sensitive settings ensure that local media resources and SRST references are used by
the roaming device. Inaddition, they ensure the correct use ofcodecs and CAC between sites.
Typically, this is always desired when adevice roams between different sites. It is not required
when the device moves only between IPsubnets within the same site. Therefore, the
recommendation isto assign all device pools that are associated with IPsubnets (device
mobility info) that are used atthe same site tothe same physical location. This action results in
phone configuration changes only when the phone roams between sites (physical locations) and
notina situation where a phone is moved only between different networks of thesame site.
Device Mobility-related settings affect call routing. By the application ofthe device CSS. AAR
group, and AAR. CSS calls can be routed differently depending on the site where the phone has
roamed to. The settings at the roaming device pool determine which gateway will be used for
public switched telephone network (PSTN) access and AAR PSTN calls (based on the device
CSS and AAR CSS) and how the number to be used for AAR calls is composed (based on the
AAR group).
Such changes can result in different dialing behavior. For instance, when roaming between
different countries, the PSTN access code might be different and PSTN numbering plans (for
example, how to dial international calls) might apply. As an example, in order to dial the
Austrian destination +43 699 18900009 from Germany, users dial 0.0043 699 18900009. while
users in the United States have to dial 9.01143 699 18900009.
4-18 Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 © 2010 Cisco Systems, Inc.
7. The roaming-sensitive settings ofthe chosen device pool (that is. the roaming device pool)
arcused to update the configuration of thephone.
Note In this case, overlapping settings (that is,settings thatexist at the phone as well as at the
device pool, namely, MRGL, Location, and Network Locale) ofthe roaming device pool have
priority over the corresponding settings atthe phone. This behavior isdifferent from the
default behavior (see Step 10).
8. Ifthe dev ice mobility groups ofthe chosen device pool and ihe home device pool are the
same, the phone configuration isupdated by applying the Device Mobility-related settings:
otherwise continue.
Note In this case, all settings are overlapping settings (that is, all Device Mobility-related settings
exist atthe phone as well as at the device pool), and the parameters of the roaming device
pool have priority over the corresponding settings atthe phone. This behavior isdifferent
from the default behavior (see Step 10).
9. Where the phone configuration has been updated (either with the roaming-sensitive settings
onh orwith the roaming-sensitive settings and the Device Mobility-relalcd settings), the
phone is reset in order for the updated configuration to be applied to the phone.
Caution This is the end of the process: do not continue to Step 10. Step 10 was directly referenced
from previous steps in certain conditions and does not apply after Step 9.
10. Here is adescription ofthe default behavior. First, the settings ofthe home device pool
(that is. the device pool that is configured at the phone) are applied. Some configuration
parameters of the device pool can also be set individually at the phone. These overlapping
phone configuration parameters are MRGL Location. Network Locale. Device Mobility
CSS (which is called simply CSS at the phone). AAR CSS, and AAR Group. Ifthese
parameters are configured at the phone (that is. are not set lo [None]), the phone
configuration settings have priority o\er the corresponding setting at the device pool.
m*
^D10 Cisco Systems. Inc implementation of Features and Applications for Mulfsite Deployments 4-17
Device Mobility Operation: Flowchart
The figure illustrates Device Mobility operation in a flowchart.
| DP - device pool. DWG =device mobility group, DMI - flevice mobility info |
— Ifit is not enabled for the device, the default behavior applies (go to Step 10);
otherwise continue.
3. Cisco Unified Communications Manager checks whether the IP address ofthe IP phone is
found inone of the device mobility groups.
— Ifit is not found, the default behavior applies (go toStep 10); otherwise continue.
4. The device pool to be used is chosen.
— Ifthe home device pool isassociated with the device mobility info in which the IP
address of the phone was found, the home device pool ischosen.
— Ifthe home device poo! is not associated wilh the device mobility info in which the
IP address ofthe phone was found, the device pool ischosen based on a load-
sharing algorithm (ifmore than one device pool is associated with the device
mobility info).
5. Ifthe chosen device pool is the home device pool, the default behavior applies (go to
Step 10); otherwise continue.
6. Ifthe physical locations of the chosen device pool and the home device pool are the same
the default behavior applies (goto Step 10); otherwise continue.
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 )2010 Cisco Systems, Inc.
In summary, the roaming-sensitive parameters are applied when the physical location ofthe
current device pool is different from the physical location ofthe home device pool (thai .s.
when roaming between physical locations). The Device Mobility-related settings are applied in
addition to the roaming-sensitive parameters when the physical locations are different and the
dev ice mobility groups are the same (that is, when roaming between physical locations within
the same device mobility group).
As a result, physical locations and device mobility groups should be used in two ways:
• Physical locations: Configure physical locations in such away that codec choice and CAC
truh reflect thecurrent location of the device. Hnsurc thatlocal SRST references and local
media resources atthe roaming site are used instead ofSRST references and media sources
that are located atthe (currently remote) home network. Depending upon the network
structure andallocation of services, vou may define physical locations that arebased upon
a cit\. an enterprise campus, or a building.
• Device mobility groups: Adevice mobility group should define agroup ofsites with
similar dialing patterns or dialing behavior. Device mobility groups represent the highest-
lev el geographic entities in your network. Depending upon ihe network size and scops,
your device mobilitv groups could represent countries, regions, states or provinces, c.ties,
or other entities. Because Device Mobility-related settings (which areapplied only v hen
roaming within the same device mobility group) affect call routing, different device
mobilitv groups should be set up whenever roaming users should not be forced to adapt
their dialing behavior. In this case, as in roaming between different device mobility groups,
the phone configuration parameters that affect call routing (that is. the Device Mobility-
related settings) arc not modified.
Note When using globalized call routing and local route groups, device mobility groups aru
irrelevant Thereason isthatthere is no need tochange thedevice-level CSS, theAAR
CSS, and the device-level AAR group. More information about the interaction of globalized
call routing and Device Mobility isprovided in a later topic of this lesson.
** ~ 10 C|sco Systems, mc implementation of Features and Applications for Muttisite Deployments 4-15
Device Mobility Operation
This topic describes how Device Mobility works.
1 Each phone is configured with a device pool (that is, the home
device pool).
IP subnets are associated with device pools,
Ifthe IP address of the phone matches a configured IP subnet
one ofthe associated device pools is selected (load-shared).
Ifthe selected device pool is different from the home device pool
of the device, these settings of the two device pools are
checked:
- If the physicallocationsare not different the phone
configuration is not modified.
- If the physical locationsare different, the roaming-sensitive
settings of the roaming device pool are applied.
- Additionally, ifthe device mobility groups are not different
the Device Mobility-related settings of the roaming device
pool are also applied.
In all other cases, the home device pool configuration is applied.
As discussed earlier, each phone isconfigured with a device pool. This device pool isthe home
device pool of the phone.
IP subnets are associated with device pools (by configuring device mobility infos).
Ifa phone for which Device Mobility isenabled registers with Cisco Unified Communications
Manager and has an IP address that matches an IP subnet that is configured in a device mobility
info, these actions occur:
— Ifthe device mobility info isassociated with one ormore device pools other than the
home device pool ofthe phone, one ofthe associated device pools ischosen based
on a load-sharing algorithm (round robin).
• Ifthe current device pool is different from the home device pool, these checks arc
performed:
— Ifthe physical locations are not different, the configuration ofthe phone is not
modified.
Implementing Cisco Unified Communications Manager. Part 2(CIPT2) v8.0 >2010 Cisco Systems. Inc.
In summarv. the U.S. device mobility group consists oftwo physical locations: San Jose and
New York.'ln San Jose. IP subnets 10.1.1.0/24. 10.1.2.0/24. and 10.1.3.0/24 arc used: New
York uses IP subnet 10.3.1.0/24. and London is configured with IP subnet 10.10.1.0/24.
Basedon the IP address of an IP phone. Cisco Unified Communications Managercan
determine oneor more associated device pool or pools and the physical location anddevice
mobility group ofthe device pool orpools. Ifan IP phone uses an IP address ofIP subnet
10.1.3.6/24. there are two candidates for the device pool. However, inthis example, the
physical location and the device mobility group are the same for these two device pools.
>201D Cisco Systems, Inc. Implementation of Features and Applications for Multisite Deployments 4-13
Relationship of Device Mobility Configuration Elements
The figure illustrates how the Device Mobility configuration elements relate to each other.
W. .> _ •. _ _ _
LON dmi
LON_ dp —• LON_pl j i GB_dmg , i
(10.10.1.0/24) (LON-campus) j
—
.
Theexample in the figure shows five device mobility infos. They are configured as follows:
• SJl_dmi: The IP subnet of this device mobility info is 10.1.1.0/24. This device mobility
info is used at Building A of the San Jose campus and is associated with device pool
SJ_A_dp.
• SJ2_dmi: The IPsubnet of this device mobility info is 10.1.2.0/24. This device mobility-
info is used at Building Bof the San Jose campus and is associated with device pool
SJ_Bl^dp.
• SJ3_dmi: The IPsubnet of this device mobility info is 10.1.3.0/24. Like SJ2_dmi, this
device mobility info isused at Building Band isassociated with device pool SJ_Bl_dp. It
is also associated with devicepool SJ_B2_dp.
• NY_dmi: The IPsubnet ofthis device mobility info is 10.3.1.0/24. This device mobility
info is used at the New York campus and is associated with device pool NY _dp.
• LON_dmi: The IPsubnet of this device mobility info is 10.10.1.0/24. This device mobility
info is used atthe London campus and is associated with device pool LON_dp.
Device pools SJ_A_dp. SJ_B1 jjp. and SJ_B2_dp are configured with the same physical
location (SJ_pl) because they are used for devices that are located at the San Jose campus.
Device pool NY_dp. serving the New York campus, is configured with physical location
NY_pl. and device pool LON_dp, serving the London campus, isconfigured with physical
location LON_pI.
All device pools that are assigned with a U.S. physical location (that is, SJ_A_dp, SJ_Bl_dp,
SJ_B2_dp, and NYjJp) are configured with device mobility group US_dmg. This setting
means that all U.S. device pools are in the same device mobility group. The London campus is
in a different device mobility group: GB_dmg.
4-12 Implementing Cisco Unrfied Communications Manager, Part 2 (CIPT2) v8.0 »2010 Cisco Systems, Inc.
Device Mobility Configuration Elements
This topic describes the configuration elements that arc used by Device Mobility.
Configuration
Configuration Element Functon
Element Name
The table lists the Dev ice Mobility-related configuration elements and describes their function,
fhe newlv introduced elements arc device mobility infos, the physical location, and the device
mobility group.
The dev ice mobilitv info isconfigured with a name and an IP subnet and isassociated with one
ormore device pools. Multiple device mobility infos can be associated with the same device
pool.
The physical location and the device mobility group are just lags: they are configured with a
name onlv and do not include any other configuration settings. Both arc nonmandatory device
pool configuration parameters: tti3t is. at the device pool, no physical location or one physical
location and one (or no) device mobility group can be chosen. They are used todetermine
whether two device pools are at the same physical location and in the same device mobility
group.
As shown in the figure, the location-dependent parameters (that is, roaming-sensitive settings
and Device Mobility-related settings)are configured at device pools. Basedon the IP subnet
that is used by the phone (which is associated with a device pool), Cisco Unified
Communications Manager canchoose the appropriate device pooland apply the location-
dependent parameters. With Device Mobility, Cisco Unified Communications Manager is
aware of the physical location of a device andapplies the appropriate location-specific
configuration by selectingthe corresponding device pool.
4-10 Implementing Cisco Unitied Communications Manager, Part2 (CIPT2) v8.0 ) 2010 Cisco Systems, Inc.
Note The physical location and device mobility group parameters are used to determine which
settings should be applied to a roaming phone (none, the roaming-sensitive settings only, or
the roaming-sensitive settings and the settings that are related to DeviceMobility). They are
not phone configuration parameters themselves, so therefore, they are not applied to the
phone configuration like the other listed roaming-sensitive settingsare. Theyare used only
at the decision to change the phone configuration and how to change it. Consequently, they
cannot be overlapping and are configurable only at device pools.
AAR CSS
— AAR Group
— Calling Party Iransformation CSS
Note All listed DeviceMobility-related settings are overlapping parameters, that is, they are
configurable at phones and at device pools. However, the Device Mobility CSSis called
CSS only inthe Phone Configuration window. Itis notoverlapping with the CSS that is
configured at the line level.
Roaming-sensitive settings are settings that do not have an impact oncall routing. Device
Mobilitv -related settings, however, may have animpact on call routing because they modify the
dev ice CSS. AAR group, and AAR CSS. Depending onthe implementation of Device
Mobilitv. roaming-sensitive settings only—or bolh roaming-sensitive settings and Device
Mobilitv -related settings—can be appliedto a roaming phone.
The GUI does not show the local route group in the roaming-sensitive settings pane.
Nevertheless, the local roule group is a roaming-sensitive setting and is updated when the
physical locations ofthe home device pool and the roaming device pool are different. The
called party transfoniiation CSS is shown in the Device Mobility-related settings pane ofthe
GUI. but this setting does not apply toIP phones and hence isno Device Mobility-related
m
setting, although shown as such in the GUI.
©2010 Cisco Systems. Inc Implementation otFeatures and Applications for Multisite Deployments
Device Mobility—Dynamic Phone Configuration Parameters
This subtopic shows the two types ofphone configuration parameters that can be dynamically
assigned by Device Mobility.
- Netw:);k Locale
- Physical Locations
Device MobilityGroup
• Device mobility-related settings:
• Dew.e Mobihiy CSS
•V.RCRS
- A/.RGi^ir.
- L?.-ii':n,) Party Ti30»fi.imsai!O!i CSS
Device Mobilitv can reconfigure site-specific phone configuration parameters that are based on
the physical location ofthe phone. Device Mobility does not modify any user-specific phone
parameters or any IPphone button settings such as directory numbers.
The phone configuration parameters that can be dynamically applied todie device
configuration are grouped into two categories:
• Roaming-sensitive settings:
— Date/Time Group
— Region
— Location
Note The Date/Time Group, Region, and Location are configured at device pools only.
m
— Network Locale
— SRST Reference
— MRGL
Note The Network Locate, SRST Reference, and MRGL are overlapping parameters. That is, they
areconfigurable at phones and atdevice pools.
— Physical Location
— DeviceMobility Group
Implementing Cisco Unified Communications Manager, Pan 2(CIPT2) v8.0 ©2010CiscoSystems. Inc.
Device Mobility Overview
This topic describes the key characteristics and Icalures of Device Mobility.
Dev ice Mobilitv can beused in multisite environments with centralized call processing. It
supports Skinny Client Control Protocol (SCCP) or Session Initiation Protocol (SIP) Cisco IP
phones andCisco IP Communicator.
Dev ice Mobilitv allows users to roam between sites with their IP phones. Typically, these are
Cisco Unified Wireless IP phones or Cisco IP Communicator phones.
When the device is added to the network ofroaming sites, itis first assigned an IP address.
Because the IP networks are difTerent for each site. Cisco Unified Communications Manager
can detemiine the physical location of(he IP phone that is based on its IP address.
Based on the physical location ofthe IP phone. Cisco Unified Communications Manage:
reconfigures the IPphone with site-specific settings.
PSTN gateways tone used are fixed. Dynamic phone CSS allows for site-Independent
local gateway access.
SRST reference is feed. SRST inference is flynamicaBy assigned.
When mobile user moves to efferent region,
codec settings are not adjusted. Region settings are dynamicallyassigned.
AAR does not work for mobile users. AAReating search space andAARflroup of
directorynumbers are dynamically assigned.
Media rescurces are assigned independent of
location. Meda resource listis dynamically assigned.
AAR causes Issues wih Cisco Extension Cisco Extension Mobtity also benefits from
Mobitty. dynamic assignment.
ITie device stil! registers with the same Cisco Unified Communications Manager cluster, but it
adapts some ofits behavior that is based on the actual site where itis located. Those changes
aretriggered by the IP subnet inwhich thephone is located. Thetable shows which issues are
solvedby Device Mobility.
Basically, all location-dependent parameters can be dynamically reconfigured by Device
Mobility. Thus, the phone keeps its user-specific configuration, such as directory number,
speed dials, and call-forwarding settings, but adapts location-specific settings like region,'
location, orSRST reference to the actual physical location. Device Mobility can also be
configured in such away that dial plan-related settings, such as the device calling search space
(CSS), AAR group, and AAR CSS, are modified.
4-6 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 »2010 Cisco Systems, Inc.
Issues with Roaming Devices
When phones move between different Cisco Unified Communications Manager sites, some
settings can become inaccurate.
The configuration ofan IP phone includes personal settings and location-dependent settings
thai are all bound statically to the MAC address ofthe phone and hence to the device itself. The
phv sical dev ice location isassumed to be constant.
Ifa phone or. more likely, asoftphone is moved between sites, the location-dependent settings
become inaccurate. Some of these settings and theirerrors arcas follows:
•m
• Region: Might cause wrong codec settings
• Location: Might cause wrong Call Admission Control (CAC) and bandwidth settings
• Survivable Remote Site Telephony (SRST) reference: Might cause malfunction ofCisco
Linified SRS I
• Automated alternate routing (AAR) group: Might cause malfunction ofthe call
redirection on no bandwidth
• Calling search spaee (CSS): Might cause usage of remote gateways instead ofloeai
gateways
• Media Resource Groups (MRGs) and Media Resource Group Lists (MRGLs): Might
cause allocation ofwrong media resources, such as conference bridges oriranseoders
To maintain the correct settings. Cisco Unitied Communications Manager needs lo be aware of
the ph> sical location ofall phones, including moving devices.
Roaming Devices
Cisco
Unified
Com municatj oris
Remote
Manager
Gateway
•s. WAN
Roaming Device
When users roam between sites, they might take their phones with them. This situation
typically does not apply lo Cisco IP phones, but is very common with softphoncs such as Cist
IP Communicator orCisco Unified Wireless IP phones.
4-4
Implementing Cisco Unified Communications Manager. Part 2(CIPT2) V8.0 >2010 Cisco Systems, Inc.
Lesson 11
Objectives
Upon completing this lesson, you will be able to describe how Device Mobility works and how
it is implemented. This ability includes being able to meet these objectives:
Identify the issues with devices roaming between sites
Describe the Device Mobility feature
Describe the Device Mobility configuration elements and their interactions
Describe Device Mobility operation
Describe Device Mobility interaction with globalized call routing
Contigurc Device Mobility
ttfc
mm
4-2
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 >20f0 Cisco Systems, Inc.
Module 41
Implementation of Features
and Applications for Multisite
Deployments
Overview
Users in multisite environments often roam between sites. They cither take devices (such as a
Cisco Unified Wireless IP Phone or Cisco IP Communicator) with them or use guest phones at
the siles that they roam to.
This module describes Cisco Unified Communications Manager Device Mobility and Cisco
Extension Mobilitv and their implementation. The implementation provides users with the
freedom to roam and still be reachable bv their own extensions, no mailer where they are or
what dev ice thev use.
Module Objectives
Upon completing this module, you will be able to implement Device Mobility and Cisco
Fxtension Mobility. This ability includes being able to meet these objectives:
• Dcscnbc how Device Mobilitv works and how il is implemented
• Describe how Cisco Extension Mobility works and how il is implemented
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 )2010 Cisco Systems, Inc.
Monitoring Leamed Routes in Cisco Unified Communications Manager Express 5-57
CCD PSTN Backup—CSS jj-58
SRST Considerations J"^
CCD and Static Routing Integration Considerations £-bi
Cisco IOS SAF Client Considerations When Using Globalized Call Routing 5-b^
Solution for PSTN Backup Advertised in E.164 Format Without Leading + 5-63
TEHO Considerations ™
Trunk Considerations When Using Globalized Call Routing ^
Cisco Unified Communications Manager Clusters and CCD Configuration Modes 5-bb
Other SAF and CCD Considerations 5-67
„ d-oo
Summary _g8
References ,„
Module Summary 5]6g
References ,. -,.
Module Self-Check r~
Module Self-Check Answer Key
ii ImplementingCisco Unified Communications Manager, Part 2 (CIPT2)v8.0 © 2010 Cisco Systems, Inc.
Table of Contents
Volume 2
Q2) A
Q-D B
Q4) C
05) D
06) B
Q-) Standard locations-based CAC docs nol allow tiic configuration of a dilTerenl limit per pair of locations.
Onh a tola! limit for all calls coming in to or going oul of a location can be configured.
Q8) B
Q9) B. [•
QIO) B
qui A
©2010 Cisco Systems, Inc. Bandwidth Management and CAC Implementation 3-109
Q7) What is alimitation ofstandard locations-based CAC? (Source: Implementing CAC)
Q8) Which statement is false about CAC when using RSVP-enabled locations? (Source:
Implementing CAC)
A) CAC adapts to the actual topology and considers network changes.
B) RSVP is used for CAC and toprovide QoS for each individual stream.
C) The RSVP agent to be used by aphone is determined by the Media Resource
Group Listof the phone.
D) The RSVP agent isconfigured as an MTP in Cisco Unified Communications
Manager.
09) AAR reroutes calls to the PSTN for which two types ofcalls? (Choose two.) (Source:
Implementing CAC)
A) calls rejected by an H.323 gatekeeper
B) calls rejected by standard location-based CAC
C) calls placed tounregistered phones
D) calls placed toa gateway that is busy
E) callsrejected by RSVP-enabled location-based CAC
F) calls rejected bySIPprecondition-based CAC
Q10) When using end-to-end RSVP with SIP preconditions, RSVP is used between the
originating and the terminating phone. (Source: Implementing CAC)
A) true
B) false
QII) How can calls that are rejected by an H.323 gatekeeper be rerouted by using adifferent
path? (Source: Implementing CAC)
A) by configuring route lists and route groups with backup devices
B) by putting the gatekeeper-controlled intercluster trunk or H.225 trunk into a
location that is set to unlimited
C) by configuring a second route pattern in the same partition that refers to a
backup device cannot be done, because AAR only supports internal calls
3-J08
implementing Cisco Unrfied Communications Manager, Part 2(CIPT2) v8.0 ©2010 Cisco Systems, Inc.
Module Self-Check
Use the questions here to review what you learned in this module. The correct answers and
solutions are found in the Module Self-Check Answer Key.
Ql) Which feature does not conserve bandwidth in the IP WAN? (Source: Managing
Bandwidth)
A) RTP headercompression
B) low-bandwidth codecs
C) local media resources
D) quality of service
02) How can the bandwidth per call be limited in Cisco Unitied Communications
Manager? (Source: Managing Bandwidth)
A) b\ specifying the maximum permitted codec bandwidth between pairs of
regions
B) bv specifying the maximum permitted codec bandwidth between pairs ot
locations
C) by specifying the maximum bandwidth per stream with the bandwidth zone
local command
D) by specifying the maximum permitted codec bandwidth tor calls going out ot
or coming into a region
Q3) When dcploving local conference bridges at each site, what is the minimum number of
Media Resource Group Lists that arc required? (Source: Managing Bandwidth)
A) number ofsites *(number ofsites - l)/2
B) one per site
C) one per site and one per conference bridge
D) one
Q4> Which device requires access to Ihe transcoder from its Media Resource Group List
when transcoding is required for acall? (Source: Managing Bandwidth)
A) both endpoinls of the call
B) the callingdevice
C) ade% ice that supports only codecs that are not permitted tor the call
D) the called device
Q5) Which statement is true about multicasl MOI 1from branch router flash? (Source:
Managing Bandwidth)
A) Multicast MOH from branch router flash requires an SRST router to be in
fallback mode to work.
B) Multicast MOH from branch router flash can also be used tor unicast MOH.
C) The branch router supports G.711 and G.729 only for MOH.
D) Regions in Cisco Unified Communications Manager have to be configured in
such away that G.711 is allowed between the Cisco Unified Communications
Manager MOH server and the branch phones.
Q6) Which CAC-related feature applies to intercluster calls? (Source: Implementing CAC)
A) locations
B) H.323gatekeeper CAC
C) AAR
D) RSVP-enabled locations
Module Summary
This module described the available design options and features that are recommended tor
deplovment in amultisite environment in order to reduce bandwidth requirements mthe II
WAN The module also described the different ways of implementing Call Admission Contro
(C \C) within acluster and beyond cluster boundaries. Finally, the module explained how calls
can be rerouted over the public switched telephone network (PSTN) ifthere is insufficient
bandwidth.
References
For additional infomiation. refer to these resources:
• Cisco Svstems. Inc. Cisco Unified Communications System 8.x SRMX April 2010.
http-,'Uuw.cisco.c<>iii'cii/"US/docs.'\oicejp_.co.nin/cucm/snid/8x/uc8x.htinl
• Cisco Systems. Inc. Cisco Unified Communications Manager Administration Guide
Release 8.0/1). February 2010. _
lmp:/^vNu.asco.coni/en.'l.S.'doc>.\oicc_ip_comin/cuem/ailn.m/8 O.J/ecmdg.bccm-801-
cm.html
• Cisco Svstems Inc. Cisco Unified SRSTSystem Administrator Guide, December 2007.
IUtp:.vw;w.cisco.coni/eivUS/partner/docs/voice ip^onitn/aisrst/adtnin/srst/configurtmc.,..
guide srr.lsa.hlml
. Cisco Svstems. Inc. Cisco IOS H.323 Configuration Guide Release 15.0- Configuring
H3"3 Gatekeepers and Proxies. February 2008. October 2009.
lu.p. x,uxv.cisco.co,n.cn/US/partner/docs/.os/voice/l,323/conngu,-at.on/gu,dc.vh.h32, gk
config psl05'Jt TSD Products Configuration Guide.Chupler.lilm!
Summary
References
For additional information, refer to these resources:
• Cisco Systems, Inc. Cisco Unified Communications System 8.x SRND, April 2010.
http:/Avuu.cisco.com/en/US/docs/voice_ip .coinm/cticm/srnd/8N/uc8\.himl
• Cisco Systems, Inc. Cisco Unified Communications Manager Administration Guide
Release 8.0(1), February 2010.
h!tp:/Av\u\.cisco.c(>m/enAJS/doLVvoicc_ip_comnVcucin/admin/8_0_l/ccmcfe/bccni-80I-
em.html
• Cisco Systems, Inc. Cisco IOS H.323 Configuration Guide Release 15.0 - Configuring
H323Gatekeepers and Proxies, February 2008, October 2009.
http://\\u^.cisco.com/en/L!S/partner/docs/ios/voicc/h323/coniiguration/guidc/vh_h323 gk
contIg_psl059i_TSD_Products_Connguration_Guide_Chapter.html
3-104 Implementing Cisco Unified Communications Manager. Part 2(CIPT2) v8.0 ©2010 Cisco Systems, Inc.
Note The command syntax and a sample configuration were shown earlier in this topic.
To provide backup paths for the gatekeeper-controlled trunk, perform this step:
Step 1 Add PSTN gateways lo route groups, and add these gateways to the route lists that
are using the gatekeeper-controlled trunks.
Note You configured PSTN backup in earlier lab activities ofthis course
Note
More information about gatekeeper configuration is provided in the Implementing Cisco
Voice Communications and QoS (CVOICE) course.
Step 4 Configure route groups, route lists, and route patterns in order to route calls that
match acertain route pattern (for example, 9.5[l2][l2].„ , for the examples that
were shown earlier in this topic) tothe gatekeeper-controlled trunk.
Note
You performed the last three steps of the preceding procedure in earlier lab activities of this
course.
3-102 Implementing Cisco Unified Communicalions Manager, Part 2(CIPT2) v8.0 ©2010 Cisco Systems, Inc.
Note For a PSTN backup, you need to perform digit manipulation msuch away that the calling
number and (more importantly) the called number are transformed to always suit the needs
of the device that isactually used. This transformation can be done atthe route list, where
digit manipulation can be configured per route group. In the example, the called number 91
511 555-1234 has to bechanged to a 10-digit number for the H.225 trunk, because the
gatekeeper is configured with area code prefixes without the long distance 1 The called
number must also bechanged to an11 -digit number if rerouting the call to the PSTN
gateway is necessary. Abetter solution would be using global transformations atthe egress
devices (H 225 trunk and PSTN gateways). In a large multisite environment or in an
international deployment, the implementation of globalized call routing would be the best
solution
Route Pattern:
9.15112J2XXXXXXX
Route Group. H.225 Trunk. Clusterl
A call that is placed to a gateway or a trunk can fail for many reasons:
• The appropriate device can be down. Timeouts occur when a call isplaced toan H.323
gateway, when an ARQ message issent toan H.323 gatekeeper, orwhen keepalive
messages are notexchanged with an MGCP gateway.
• Communication problems can occur with the gateway. H.323 messages can be sent tothe
IP address of thewrong interface. Gatekeeper registration can fail because of an invalid
zone name orbecause the call isrejected due toa lack ofresources. Acall might be
rejected when no channel is available on an EI or Tl trunk, when an administratively
configured limit ofcalls isreached atadial peer, orwhen a call isdenied by CAC.
Cisco Unified Communications Manager uses the same backup method—route lists and route
groups—for all ofthese types ofcall failures. Ifthe currently attempted device ofaroute group
cannot extend the call (for whatever reason), Cisco Unified Communications Manager will try
the next device according to the route group and route list configuration.
Therefore, providing abackup for calls that have been rejected due to H.323 gatekeeper CAC is
as simple as having aroute list and route groups that prefer the gatekeeper-controlled trunk
overoneor more PSTN gateways. If thecall cannot besetup overthe trunk, Cisco Unified
Communications Manager will reroute the call to the PSTN gateways. Instead ofreferring to a
dedicated PSTN gateway that should be used as abackup, the local route group feature can be
used.
3-100 Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 52010 Cisco Systems, Inc.
The maximum audio bandwidth is limited to 64 kb/s. Calls requiring more bandwidth {for
media such as wideband audio codecs or video calls with a video call bandwidth of more
than 64 kb/s) are not permitted in any zone.
Thetotal of all calls(interzone andintrazone calls) in zone ClusterB must notexceed 688
kb/s. As anexample, ibis configuration allows three G.729 calls to ClustcrA (three times
twice the codec bandwidth of 8 kb/s) and five G.711 calls within ClusterB (five times twice
the audiocodec bandwidth of 64 kb/s). Intrazone calls in zone ClusterA are unlimited.
Note Some ofthe bandwidth commands inthe example are forillustration only andare not useful
inthis scenario. Forexample, youcould change the bandwidth interzone default 64
command to bandwidth interzone ClusterA 64 because interzone default appliesonly to
zone ClusterA; zone ClusterB is explicitly configured, and no other zones exist. Furthermore,
intrazone limitations have been configured but would never apply inthis scenario. The
reason is that all calls are interzone calls.Thegatekeeper is used only forintercluster calls,
and the two clusters are in different zones.
mm
192.168.3.254
10.1 1.1
pow Trunk: Cluster^ ^mm^ Ji.225 Trunk: CL-^ 10.1.1.1
gatekeeper
ions local ClusterA lab.ccm 192.168.3.254 10.2.1 2
10.1.1 2
ions local ClusterB lab.cm 192.169.3.254
•one prefn ClusterA 511-
H 225 Trunk'Clusterl
lone prefix ClusterA 521" H.225 Trunk; Cluster2
DeviceType. Gateway
ions prefin ClusterB 513« DeviceType.Gatenvay
Zone ClusterA
rone prefix ClusterB 522* Zone1 ClusterB
Technology Prefix Mr
bandwidth interzone default a Technology Prefix: Mr
GK 192 158 3 254
bandwidth intertone sons CI ustsrB IS GK 192.168 3.254
bandwidth session default 128
bandw4.dth total zone ClusterB 688
9"-type-prefix lt« default- technology
no shutdown
The example is based on the previously illustrated example, but now the H.323 gatekeeper also
performs CAC.
The bandwidth interzone default 64 command specifies that 64 kb/s is permitted for calls
going out of and coming into azone. Because no specific zone is specified but the keyword
default is used, this setting applies to all zones that are not explicitly configured with a
different setting.
The bandwidth interzone zone ClusterB 48 command specifics that the previously configured
default interzone bandwidth limit should not apply to ClusterB but that ClusterB should instead
be limited to 48 kb/s.
The bandwidth session default 128 command limits the bandwidth to be used per call to a
codec that does not require more than 64 kb/s (for example, G.711 or G729) Because no
d.fferent session bandwidth is configured for any specific zone, this default applies to all zones.
The bandwidth total zone ClusterB 688 command limits all calls of ClusterB (that is calls
within the cluster and intercluster calls) to atotal of688 kb/s. Because there is neither a
bandwidth total default command nor aspecific bandwidth total command for ClusterA
ClusterA has nototal limit applied.
IfG.729 is used for inter/one calls and G.711 is used for intrazone calls, this configuration
effectively would permit these limitations:
> There can be amaximum of three G.729 calls between ClusterA and ClusterB because
CusterB is limited to 48 kb/s (that is, three times twice the codec bandwidth of8kb/s)
ClusterA could have four G.729 calls to other zones. However, because the example shows
only two zones and the other zone (ClusterB) is limited to three G.729 calls ClusterA will
never beable to use thepermitted interzone bandwidth.
3-98
Implemenling Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
©2010 Cisco Systems, Inc.
Bandwidth limitations are configured differently on different Cisco products and for different
features. The table summarizes how to configure bandwidth limitations in Cisco Unified
Communications Manager.
Manager Manager
Region Location
Note
Video calls have not been discussed in this course but are also shown for completeness.
In Cisco IOS Software, you implement H.323 gatekeeper CAC by using the bandwidth
command.
Syntax Description
interzone Specifies the total amount of bandwidth for H.323 traffic from the zone to any
other zone. '
total
Specifies the total amount ofbandwidth for H.323 traffic that isallowed in the
zone.
session
Specifies the maximum bandwidth that isallowed for a session in the zone
default Specifies the default value for all zones.
zone
Specifies a particular zone.
zone-name Names the particular zone
bandwidth-size Maximum bandwidth. For interzone and total, the range isfrom 1to
10,000,000 kb/s. For session, therange isfrom 1to 5000 kb/s.
The bandwidth that is calculated per call is twice the bandwidth of the audio codec AG729
call consumes 16 kb/s of the configured bandwidth, and a G.711 call consumes 128 kb/s of
the configured bandwidth.
3-96
Implementing Cisco Unified Communicalions Manager, Part 2(CIPT2) vB.O >2010 Cisco Systems, Inc.
Thev all use different H.323 IDs because different trunk names have been contigured in the two
clusters and because Cisco Unified Communications Manager adds the _1 and _2 to the trunk
name to uniquely identify the call-processing servers per cluster.
Note If the same trunk name was configured in the two clusters, registrations would fail because
ofduplicate H.323 IDs.
The call-processing servers of ClusterA registered in /one ClusterA. and the call-processing
servers of ClusterB registered in zone ClusterB. You can verify this situation by using the
command show gatekeeper endpoints. All endpoints are registered with the prefix 1# which
is configured to be the default technology prefix. You can verify this situation by using the
command show gatekeeper gw-type prefix. The output of these two commands is shown m
one table in the figure.
Note For more information regarding gatekeeper configuration and operation, refer to the
Implementing Cisco Voice Communications and QoS (CVOICE) course. ____
Ifthe gatekeeper receives an Admission Request (ARQ) message from one ofthe H.323
aatewavs (a call-processing Cisco Unified Communications server, in this case), it looks up its
call-routing table (list of configured /one prefixes) to find out in which /.one the requested
prefix can be found.
You can verifv the list of configured prefixes and their /ones by using the command show-
gatekeeper zone prefix.
If an ARQ message was sent from 10.1.1.1 to the gatekeeper that requests acall to
512555P34 the gatekeeper will determine that the call has to be routed to zone ClusterB. Inc
onlv prefix that is reg.stered by gateways in this zone is 1#*. which is the default technology
prefix and is registered bv 10.2.1.1 and 10.2.1.2. Therefore, the gatekeeper chooses one of these
two eatcwa% s(in round-robin fashion) to be the terminating gateway. It will inform the
originating gatewav (the call-processing server ofClusterB that sent the ARQ message) to set
up an 11.323 call with the determined terminating gateway (10.2.1.1 or 10.2.1.2).
Note At this point, the gatekeeper is configured only to perform call-routing address resolution. It
resolves adialed number to the IP address where the call has to be routed. No CAC is
performed by the gatekeeper in this example
gatekeeper
zone local ClusterA lab.con
zone local ClusterB lab.com 10.2.1.2
10.1.1.2
zona prefix ClusterA 511*
H 225 Trunk Clusleri zone prefix ClusterA 521-
zone prefix ClusterB 512* H.225 Trunk: Cluster2
DeviceType Galeway DeviceType. Gateway
Zone CluslerA zone prefix CluaterB 522*
Zone: ClusterB
Technology Prefix If g»-type-prefix 1#« default-technology
no shutdown
Technology Prefix MT
GK. 192168.3254 GK. 192 16S.3.254
GATEKEEPER BKDPOINT HEGISTRATIOH
H323-ID IPAddr ZonaName Type Prefii
Clusterll 10.1.1.1 ClusterA VOIP-GW 1#*
Clusterl_2 10.1.1.2 ClusterA VOIP-GW 1#'
Cluster2_l 10.2.1.1 ClusterB VOIP-GW If
ClusterJ_2 10.2.1.2 ClusterB VOIP-GW 1#*
In the example, two Cisco Unified Communications Manager clusters arc shown Each cluster
hasan H.225 trunk configured.
The 11.225 trunks use different names per cluster in order to keep the H323 IDs unique
ClusterA uses Clusterl. and ClusterB uses Cluster2. In each cluster, there are two call-
processing nodes (10.1.1.1 and 10.1.1.2 in ClusterA, and 10.2.1.1 and 10.2.1.2 in ClusterB).
The trunk in ClusterA with the name Clusterl is configured with zone ClusterA and technology
prefix 1# . The trunk in ClusterB with the name Cluster2 is configured with zone ClusterB and"
mVITi6^!??010^ PrCfiX ('**}" B°th tnmks refer t0 **1P address oUhc sa™ gatekeeper:
I yl. 168.3.254.
The gatekeeper has two local zones: ClusterA and ClusterB. It is configured to route calls to
prefixes 511 and 521 to zone ClusterA, and calls to prefixes 512 and 522 to zone ClusterB In
addition, the gatekeeper is configured to use 1#* as the default technology prefix 'fhat is calls
to prefixes for which the gatekeeper does not know which gateway to use are routed to the
gateway orgateways that registered a technology prefix of I#*).
This gateway configuration means that the gatekeeper will have four gateways registered:
' Manager
MUSterL^*Group
WhiCuthatfS isconfigured
tHe ^ cal|-Processing server of the Cisco Unified Communications
in the device pool ofthe trunk
• Cluster ]_2. which is the second call-processing .server ofClusterA
• Thetwo call-processing servers of ClusterB
3-94
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) vB.O ©2010 Cisco Systems, inc.
Note The H.323 ID has to be unique. Cisco Unified Communications Manager keeps the H.323 ID
that is used by the members of acluster unique by adding the individual ending _x
Furthermore because Cisco Unified Communications Manager does not allow multiple
trunks to use the same name, no duplicate H.323 IDs can be presented to the gatekeeper
from a duster However, if the same trunk name is configured in multiple clusters, the call-
processing servers of two or more clusters will try to register with the same H.323 ID. The
gatekeeper will not allow these duplicate H.323 IDs to register, so the trunk will not be
operational Therefore, it is important to use unique trunk names across all Cisco Unified
Communications Manager clusters that register with a gatekeeper. _
II 323 zone: H323 zones are used to group devices. You perform call routing and CAC
based on these /ones. For instance, you could configure aso-called detault technology
prefix per zone that identifies the gateway (or gateways) to which calls should be routed
when the gatekeeper does not know which gateway to use. Also. CAC can be configured
differentlv for calls within a zone versus interzone calls.
Note The H323 zone name that is configured at the gatekeeper-controlled trunk is case-sensitive
and hastoexist atthe gatekeeper ^ __
Note More information about how agatekeeper routes calls is provided in the Implementing Cisco
Voice Communications and QoS (CVOICE) course. _
Cisco Unified Communications Manager can connect to other Cisco Unified Communications
Manager clusters or to any other H.323 devices via H.323 trunks. H.323 trunks can be
configured on their own—without the use of a gatekeeper for address resolution and CAC—or
as gatekeeper-controlled trunks. You can configure two gatekeeper-controlled trunks in Cisco
Unified Communications Manager:
• Gatekeeper-controlled intercluster trunk: This trunk is used to connect to Cisco
CallManager versions earlier than 3.2.
• H.225 trunk: This trunk can be used to connect to Cisco Unified Communications
Manager Version 3.2 or later and to all other H.323 devices. The H.225 trunk features a
peer discovery mechanism and hence can identify the device that is located at the other end
of the trunk and use the appropriate feature set.
3-92 Implementing Cisco Unitied Communications Manager. Part2 (CIPT2) vB.O 12010 Cisco Systems, Inc.
Step 2b: Apply SIP Profile to Trunk
The figure shows you how to applv the previously configured SIP profile to a trunk.
fl--!lei!=;^'.:».C.Jc-.' ':iul«» j
jr "
:..,ln-(i,-:u5* ii*r3j (1 P tara g-cw: / r
;:=-Tj-kS«:u--. f-tfw' --'IMSele;tM •- /jr V
^^jr, K-.f-
At the SIP trunk, set the SIP profileto the profilethat you createdearlier.
When the RSVP OverSIPparameter of the SIPprofile is set to Local QoS, or fall back to local
RSVP is enabled at the SIP profile, the SIP trunk needs to have an MRGL assigned sothatit
can allocate an RSVP agent for intracluster RSVP-enabled CAC. You can setthe MRGL
directK at the SIP trunk configuration page. If it is notsetat the trunk, you must setthe MRGL
at the device pool that is applied to the SIP trunk.
SetSIPRellXXOptbns
to Send PRACKif Ixx
Contains SDP
The necessary configuration for SIP Preconditions is applied to SIP trunks via SIP profiles. At
the SIP profile, you have to set the SIP RellXX Options parameter to Send PRACK if Ixx
Contains SDP.
Then you have to set RSVPOver SIP to E2E (end-to-end) whenyou want to enable SIP
Preconditions. If you want the trunk to use local QoS only, you wouldset the parameter to
Local QoS instead of to E2E.
WhenSIP Preconditions is configured (RSVPOver SIP is set to E2E),you can check the check
box fall back to local RSVP. This option allows a fallback to local QoS if the far end does not
support SIP Preconditions. If SIP Preconditions is supported by the far end and the RSVP
reservation fails, there is no fallback to local RSVP.
Note When the other side ofthe SIPtrunk is CiscoUnified Communications Manager, there will
never be a fallback to local RSVP. SIP Preconditions is never considered to be unsupported
between two Cisco Unified Communications Manager clusters, regardless whether it has
been enabled at the other side or not. As a consequence SIP Preconditions always fails and
never falls back to local RSVP in such a scenario.
When the other side of the SIP trunk is Cisco IOS device—for example, Cisco Unified
Communications Manager Express—and end-to-end RSVP is not enabled at that remote
router, then a fallback to local RSVP is performed, if configured at the local Cisco Unified
Communications Manager cluster. Ifend-to-end RSVP is not configured on a Cisco IOS
device, SIP Preconditions is considered to be unsupported and therefore local fallback is
possible.
3-90 Implementing CiscoUnified Communications Manager, Part 2 (CIPT2) v8.0 12010 Cisco Systems, Inc.
SIP Preconditions Configuration Procedure
This subtopic describes the SIP Preconditions configuration procedure.
Note The RSVP agentthatis associated with the IPphone is usedfor the call leg to the far-end
SIP device. IfQoS fallback is not enabled, the SIP trunkwill never allocate an RSVPagent.
IfQoS fallback mode is enabled, two local RSVP agents are required in a fallback scenario:
one forthe IP phone and one forthe SIP trunk. Therefore, the MRGL at the SIP trunk is
required only for QoS fallback mode orfor when theSIP trunk is not configured for SIP
Preconditions at all but is configured to use local QoS.
Thefirst configuration step was described earlier inthe lesson and is notdescribed again.
Refer to the"Configuration Procedure for Implementing RSVP-Enabled Locations-Based
CAC" subtopic inthis lesson for a description of Step 1.
If QoS fallback or local QoS configuration, the policies that are applied to local QoS are
managed the same way that they are managed for intracluster calls with RSVP-enabled
locations.
3-88 lrr.p)ementmg Cisco Unified Communications Manager, Part2 (CIPT2) v8.0 ©2010Cisco Systems, Inc.
Fallback from End-to-End RSVP to Local RSVP
This subtopic describes the fallback mechanism from end-lo-end RSVP to local RSVP when
the far end of the SIP trunk does not support SIP Preconditions,
You can configure QoS fallback to use local RSVP when end-to-end RSVP is notsupported by
the farend. fallback applies only in the casewhere the far enddoesnolsupport SIP
Preconditions. If il does supportSIP Preconditions and the RSVP reservation fails, there is no
fallback to local RSVP.
If there is QoS fallback, the call is reattempted without SIP Preconditions. CAC reverts to local
RSVP. uhich means that two cluster-internal RSVP agents arc used. The call is split into three
local call legs:
• One from the originating phone to its RSVP agent
• One from that RSVP agent to the RSVP agentthat is associated with the SIPtrunk
• One from the RSVP agent that is associated with the SIP trunk, toward the othercall-
mm
routing domain (where the same action con happen inthe ease ofa Cisco Unified
Communications Manager)
However, the call leg between the two clusters or between the local cluster and the SIP device
on the other end does not use RSVP-based CAC.
The configured RSVP policy determines how calls areprocessed in certain scenarios:
• When the far end does not support preconditions and QoS fallback is off, the call fails
when the RSVP policy is Mandatory, or Mandator,' (Video Desired). When the RSVP
policy is Optional (Video Desired), the call continues without RSVP.
• When the farend does notsupport preconditions and QoS fallback is on,the configured
RSVP policy is applied to local RSVP.
>2013 Cisco Systems, Inc Bandwidth Management and CAC Implementation 3-87
SIP Preconditions Call Flow Summary
(Cont.)
Answ3witti SlnflteCode
NegotiatedbyMedia
Whenthe call is answered, the terminating side sends an OK messagethat is confirmed from
the other side with an ACK message.
Now. where the call is formally set up. the terminating side triggersa renegotiation of media
capabilities with an INVITE message with no SDP attached.
Theoriginating sidesends an OK message, including the capabilities of the enddevice, in its
SDP.
The terminating side selects a codec and informs the originating side with an ACK message
with anattached SDP, including theselected capabilities (codec, packetization size, and soon).
Ifneeded. RSVP reservations areupdated between the two RSVP agents.
3-86 Implementing Cisco Unified Communications Manager, Part2 (CIPT2) vS.O ©2010 Cisco Systems, Inc.
SIP Preconditions Call Flow Summary
This subtopic illustrates a summarized call flow forSIP Preconditions calls.
QoS "Preconditon"
m=amJio 20O0G RTPfAVP 0
0=!NlP4 192022
m=auara10000 rtf/wP 0 a=wjrE.flose2e '.vi'-t
e=MIP4 1920 21 > a=desqos mandatory e2a
a=currqose2ei
^H.II.J.I Ii>Kiaa*gaa^ai
<
sendrecu
a=des qos mandatory e2e a^conl qos e2e recv
flendrecv
SIPUA1 initiates
RSVP reservation in SIP UA2 initiates
the 1 -» 2 direction RSVP reservation in
the 2 ^ 1 direction.
nvauOio 10000 RTPNWP 0
c=!NIP4 192021
a^currqos s2e';e--.i m=audio 20000 RTP/AVP 0
allies qos mandatory e2e C=INIP4192 02 2
sendiecv 7 a=curr qtra e2f ser **•;•.•
aides' qos mandalory e2e
•^f—BJSIfriJt'Jii'lWI
<i sendrec*<
"fhe figure shows the most important components ofthe first phase ofthe call setup over a SIP
trunk that iseonligured for SIP Preconditions. The phase starts with the initial INVITE
message with the IP address of the originating RSVP agent and the request for RSVP CAC in
the SDP.
Then it show s the 183 response message that confirms the received RSVP CAC request inits
SDP. The SDP furtherincludes the IP address of the terminating RSVP agent and the request
for RSVP CAC for the reverse direction.
This negotiation isthen completed by the PRACK message that issent from the originating
side toward the temiinating side.
RSVP reser\ ations arethen setup In each RSVP agent for thedirection to theother RSVP
agent, using RSVP PATH and RSVP Resv messages.
The originating side then informs the temiinating side about the successful RSVP reservation in
the SDP of an UPDATH message. The terminating side confirms this information in anOK
message with SDP that includes the same status information for the other direction.
The precondition phase is now completed, and the terminating device can now send a
RINGING message to the originating side.
6. When the call isanswered, the terminating Cisco Unified Communications Manager
requests a renegotiation of media capabilities by sending a SIP INVITE message without
SDP.
7. The originating Cisco Unified Communications Manager responds with aSIP OK message
with SDP. Thecomplete set of supported media capabilities is included in the SDP.
8. The receiving CiscoUnified Communications Manager sends a SIPOK withSDP
message, including the selected codec. This codec is nowactually usedfor the end-to-end
call.
9. Ifthe selected codec has bandwidth requirements that are different from the requirements
that were used during the SIP Preconditions phase, the RSVP reservation is updated
accordingly.
10. The call is now established with three call legs (like with RSVP-enabled locations for calls
within a cluster):
— The call leg between the originating IPphone and its RSVP agent, where no RSVP-
based CAC was performed
— The middle call legbetween thetwo RSVP agents, where RSVP-based CAC was
performed, as described earlier
— The call leg between the terminating IP phone and its associated RSVP agent, where
againno RSVP-based CAC was performed
Note Standard locations-based CAC is performed between the IP phones andtheir associated
RSVP agents As a result, thecall leg from theIPphone to itsRSVP agent iscounted
against themaximum bandwidth thatisconfigured at thelocations thatare applied to theIP
phone and to the RSVP agent.
3-84 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 ) 2010 Cisco Systems, Inc.
SIP Preconditions Operation
This subtopic describes the operation ofSIP Preconditions inCisco Unified Communications
Manager.
A call with SIP Preconditions follows the message sequence of RFC 3312 to establish a
precondition. Here isa summarv' ofthe session establishment phases:
1. The originating IP phone places a call to adestination that isreachable through a SIP trunk.
According to the location configuration at the originating IP phone location, RSVP has to
beused between the location ofthe originating IP phone and the SIP trunk where the call
mm
should be routed to.
2, The originating Cisco Unified Communications Manager sends a SIP INVITE message
with Session Description Protocol (SDP). The IP address for the media stream in the SDP
is setto the IP address of the originating RSVP agent. RSVP is requested in the SDP.
3 The tenninating device—for example aCisco Unified Communications Manager server of
another cluster—responds with a SIP SESSION PROGRESS message with SDP. Itwill
provide the IP address of the terminating RSVP agent, confirm the RSVP request for the
forward direction, andsend an RSVP request forthe reverse direction.
4. The negotiation of SIP Preconditions for RSVP CAC is completed by SIP PRACK and OK
messages. Then each of the two RSVP agents attempts an RSVP reservation tor i+s forward
direction (that is. toward the other RSVP agent) ofthe preconditioned bandwidth.
5. Ifthe RSVP reservation issuccessful, a standard call setup isperformed by SIP RINGING,
OK. and ACK messages.
mtw
For SIP calls going out of the cluster, RSVP can be used end-to-end
between different domains.
SIP Trunk
SCCP
RTP
• RSVP
_ Anarogor
Digital Voice
When both endsof a SIPtrunk support SIP Preconditions and the IP phone and the SIPtrunk
are in different locations and RSVP is enabled between thesetwo locations, then end-to-end
RSVP is used. Asa result, only the RSVP agent thatis associated with the IP phone is invoked;
there is nosecond local RSVP involved. TheRSVP agent of thephone now uses RSVP-based
CAC toward the other end of the SIP trunk.
Iftheother endis another Cisco Unified Communications Manager cluster, then the same result
happens at that farend: only one RSVP agent is invoked. If the otherend is a CiscoIOS router,
then that router (either Cisco Unified Communications Manager Express or a Cisco IOS SIP
gateway) terminates RSVP at the far end.
With SIP Preconditions, RSVP is now virtually end-to-end. Itspans the two call- routing
domains and is not limited to the local cluster.
Note Due to proprietary extensions, SIP Preconditions for RSVP-enabled CAC iscurrently
supported only between Cisco Unified Communications Manager, Cisco Unified
Communications Manager Express, andCisco IOS SIP gateways. Third-party SIPdevices
are currently not supported.
3-82 Implementing Cisco Unified Communications Manager, Part2 (CIPT2) v8.0 ) 2010 Cisco Systems, Inc.
CAC Without SIP Preconditions
This subtopic describes how you can implement local RSVP-based CAC for SIP trunks when
SIP Preconditions is not used.
For SIP calls going out ofthe cluster, RSVP can be used only
between local RSVP agents associated with phone and SIP
trunk.
When not using SIP Preconditions, you can use RSVP only within the local Cisco Unified
Communications Manager cluster. Such an implementation islike RSVP-enabled locations, as
discussed in an earlier topic ofthis lesson, except that the two devices that are involved in the
local Cisco Unified Communications Manager cluster arean IP phone and a SIP trunk (or two
SIP trunks).
The IP phone and the SIP trunk are in different locations, and RSVP is enabled between these
two locations. The IP phone refers to its RSVP agent by its MRGL, and the SIP trunk refers to
its RSVP agent b> its MRGL. RSVP CAC applies between these two RSVP agents. Because all
devices are local 'to the Cisco Unified Communications Manager cluster, this implementation
model is called local RSVP. Ifanother Cisco Unified Communications Manager cluster is atthe
other end of the SIP trunk, local RSVP can beused also at that end. However, the call leg
between the two RSVP agents that are associated with the SIP trunk ateach cluster isnol
subject to RSVP. Therefore, there isno end-to-end RSVP in this ease.
Ifthe other end ofthe SIP trunk isa third-party device, a Cisco IOS SIP gateway, orCisco
Unified Communicalions Manager Express, then local RSVP applies only tothe end ofthe SIP
trunk where Cisco Unitied Communications Manageris used.
Another term that is used to refer to SIP Preconditions is "end-to-end RSVP." This term does
not mean that RSVPis implemented in the actualendpoints (IP phones),but it refers to
intercluster calls. Before SIP Preconditions, intercluster callsusing SIPwereableto useonly
local RSVP within a cluster. In thiscase, an RSVP agent that is associated with the IP phone,
and another RSVPagent that is associated with the SIP trunk, are used. Sucha configuration
requires the phone and the trunk to be in separate locations, and RSVP needs to be enabled
between these two locations. Thesetwo RSVPagents,however, were both local to the Cisco
Unified Communications Managerclusterand hence were not spanningto the other end of the
cluster. With SIP Preconditions, RSVP can be used between both ends of the SIP trunk; hence
the name end-to-end RSVP.
SIP Preconditions is not limited to intercluster trunks (that is, calls between two Cisco Unified
Communications Manager clusters). It can be used also for SIP trunks to Cisco Unified
Communications ManagerExpress. Cisco IOS gateways, and Cisco Unified Border Elements.
3-80 Implementing CiscoUnified Communications Manager. Part 2 (CIPT2) v8.0 ) 2010 Cisco Systems. Inc.
Step 4: Configure Phones for AAR (Cont.
Forward calls to
voce mail if directory Set individual destination number
Set (verify) exiemal
number cannot be mask (CFNB) to be used instead of
phone number mask
reached due lo external phone number mask and
otdireclory numbef
locai ms-based AAR group prefix
CAC.
You need lo configure the IP phone directory numbers for AAR. The Directorv- Number
Configuration windowdisplays these relevantoptions:
• Voice Mail: If this check box is checked, calls to this phone are forwarded to voice mail if
this directory numbercannotbe reached due to locations-based CAC.
• AAR Destination Mask: If thisoption is set. the number where callsare rerouted to if this
directory number cannot be reached due to locations-based CAC iscomposed of this mask
and this director, number. Otherwise, the number would be composed of this directorv
number, the external phone number mask, and an AAR group prefix. Because 1his setting is
configured for each directory number, it allows any destination to be specified. (Ifthere arc
not n wildcard digits inthe mask, then calls arererouted to the specified number without
considering any digits ofthe directorv' number.) Therefore, this setting isoften referred lo
as CFNB.
• AARGroup: An AAR group at the directory number has to be set in orderto allow AAR
calls to thisdirector.' number. The AAR group that is configured at thedirectorv number is
the destination AAR group.
• External Phone Number Mask: This mask is the external phone number mask of the
directory number. Itshould always be set, because it is used by other features (such as digit
manipulation at route patternsor route lists).
When you enable AAR on a phone, there are two possible settings in the Phone Configuration
window:
• AAR CSS: This CSS is used ifa call that originated at this phone is rerouted using AAR.
• AAR Group; The AAR group of the phone is the source AAR group, while the AAR
group that was set at the directory numberis the destination AAR group. It is important to
understand this distinction for the configuration of AAR prefixes,becausethey are
configured separately foreachpairof AAR source anddestination group. If no AAR group
is set at the phone, then the AAR group of the directory number is used as the AAR source
group for this phone.
3-78 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 12010 Cisco Systems, Inc.
Step 3: Configure AAR Groups
As mentioned before. Step 2. "Configure partitions and CSSs." is not shown, because the
configuration of partitions and CSSs has been covered earlier. This figure illustrates Step 3. the
configuration of AAR groups.
You configure AAR groups from Cisco Unified Communications Manager Administration
underCall Routing > AAR Groups. Each addedAAR group can be configured with a dial
prefix for itsown group and twodial prefixes foreachof the otherAARgroups (one forcalls
going to theother group andone forcallsbeingreceived from theothergroup).
Note Inthis example, there are onlytwoAAR groups. For AAR calls from HQto BR a prefix of
0001 is used. For calls tn the other direction, a prefix of 901149 is used. The AAR
configuration that is shown would fit to a scenario where the HQ site is in Germany and the
BR site is in the United States The external phone number mask at both sites would use
national format.
As a result, an AAR call from Germany to the United States would be placed to 0001
followed by the national number (10 digits). 0 is the PSTN access code in Germany, 00 is
the international access code, and 1 is the country code for the United States. An AAR call
from the United States to Germany would be placed to the national number of a Germany
phonethat is prefixed with 901149. 9 is the PSTN access code inthe United States, 011 is
the international access code, and 49 is the country code of Germany.
Tip The configuration thatisshown in thefigure does not useglobalized call routing. Globalized
call routing is recommended in larger multisite environments, especially ininternational
deployments. With globalized call routing, all sitesuse thesame AAR group and noprefixes
are required within thatgroup The external phone number mask is specified in globalized
format(E.164 number with + prefix).
) 2010 Cisco Systems. Inc. Bandwidth Management and CAC Implementation 3-77
Step 1: Configure AAR Service Parameters
The figure illustrates how to enable AAR and how to set AAR-related parameters.
You enable AAR by setting the Cisco CallManager service parameter Automated Alternate
Routing Enable to True (False is default).
Other AAR-related service parameters aretheOut-of-Bandwidth Text parameter, where you
can set the text that is displayed on an IP phone when a call fails due to no available bandwidth,
and the AAR Network Congestion Rerouting Text parameter, whereyou can set the text that is
displayed on an IP phone when AAR reroutes a call.
3-76 Implementing CiscoUnified Communications Manager, Part 2 (CIPT2) v8.0 >2010 Cisco Systems, Inc.
AAR Configuration Procedure
The implementation of AAR includes these steps.
Step 2.-Configure partitions and CSSs." is notdiscussed in this topic, because the
configuration of partitions and CSSs was discussed in detail in the Implementing Cisco Unified
Communications Manager, Part 1 (CIP'1'1) course,and both of these itemshave already been
used se\eral limes in this course.
You need to precisely design partitions and AAR CSS. The AAR CSS of the calling device
must include the partition that is necessary to route the redirected call. Thecall is routed to the
number that iscomposed of the destination directory number, external phone number mask, and
AAR prefix (according tothe AAR group configuration). Ifyou configure an individual AAR
destination mask or forward to voice mail, the AAR CSS has to provide access to these
numbers (numbers that arccomposed of the called directory number and AAR destination mask
or voice-mail pilot number).
As mentioned earlier, in globalized call routing, AAR configuration is simpler when you use
the globalized format at theexternal phone number mask.
AAR Considerations
• Call originates from an IP phone within one location and terminates at an IP phone within
another location.
• Incoming call through a gateway device within one location terminates at an IP phone
within another location.
AAR does not work with Survivable Remote Site Telephony (SRST). AAR is activated only
after a call is denied by CAC, not by WAN failures.
Using globalized call routing simplifies the implementation of AAR substantially—especially
in international deployments.
AAR does not support CTI route points as the origin or destination of calls, and AAR is not
compatible with Cisco Extension Mobility for users who roam to different sites.
Note When tail-end hop-off (TEHO) is used, itisimportant toconfigure AAR in such a way that
the local gateway is always used for callsbeing rerouted by using AAR. This automatically
occurs when you use local routegroups. When youare not using local routegroups, you
have to configure AAR CSS so that the focal gateway is used forAAR calls. Ifthe AAR CSS
refersto the TEHO gateway, AAR callswill fail, because the callleg to the (remote) PSTN
gateway again has the same issue that the initial call had: Itneeds to go overthe IP WAN
(which typically meansthat itgoes outofthe location ofthe originating phone), butdoing
thatis not possible becauseno bandwidth is left for the location (which was the reason why
the initial called ended up in a CACfailure).
3-74 implementing CiscoUnified Communications Manager, Part2 (CIPT2) v8.0 32010 Cisco Systems, Inc.
In the example, all phones are in the same AAR group (System). Noprefix is configured for
callswithin thissingle AAR group. There is a single route pattern in Ii.164 format: (\+!). The
route pattern refers to the only configured route list, which is configured to usethe local route
group. Each gateway is referenced from a site-specific route group. U.S. phones use a U.S.-
specific device pool with the local route group setto U.S., and German phones use a device
pool specific to theircountry. where the local route group refers to the DE route group. The
external phone number mask in globalized format is +15115222xxx at U.S. phones and
+4969125xxxx at Gemian phones. The AAR CSS is the same for both phones and provides
access to the '*+.! route pattern.
When a call from a U.S. phone to a German phone is notadmitted because of no available
bandwidth, theexternal phone number mask of the German phone is merged withthe directory
number ofthe phone (in thiscase, the result is +49691253001). No AAR prefix is added, so a
call isplaced to that number. It matches the \+.! route pattern, and the local route group isto be
used, fherefore. the call is sent to the U.S. gateway, where the called numbercan be localized,
using called-party transformation (that is. the number ischanged lo49691253001 with a
number type of international) settings thatare configured at the gateway.
The same thing happens for calls in the other direction. As a result,+15115552001 is called,
and after the called numberis localized at call egress—again provided by global
Iransformations at the gateway—a call with a number type of international is placed to
15115552001. this time through the Gennan gateway.
©2010 Cisco Systems. Inc Bandwidth Management and CAC Implementation 3-73
AAR Example with Local Route Groups and Globalized
Numbers
The figure shows an AAR examplewhere globalized call routing and local routegroups are
used.
,. Sitel Site 2
IP WAN
PSTN
Route PaBem;**!
1
Single Roule List
Default Local Route Group
DN" 2001 DN": 3001
Ext Phone Ext. Phone
Number Mask
Single Route Pattern in One
Number Mask
+ 15115552XXX
Partition, Single AAR CSS, Single
+4969125XXXX
AAR Group, No Prefixes Within AAR
Group
If the AAR destination mask is entered in the globalized form, and if every AAR CSS is able to
route calls to destinations in the globalized form, then system administrators can forego the
configuration of AAR groups, because their sole function is to determine which digits to prefix
based on the local requirements of the PSTN access of the calling phone to reach the specific
destination. With globalized call routing, Cisco Unified Communications Manager can route
calls to the PSTN in E. 164 format with a + prefix. When you configure the external phone
number mask, in this format, no prefixes are required for AAR. To localize the called- and
calling-party numbers, implement global transformations for each egress PSTN gateway (like
for normal PSTN calls).
Without local route groups, the AAR CSS is used to route the call through the colocated
gateway of the calling phone by matching a site-specific route pattern that refers to a site-
specific route list, route group, and gateway. When local route groups and globalized call
routing are implemented, the egress gateway does not need to be selected by site-specific AAR
CSS. because the egress gateway is determined by the local route group feature.
In summary. \\ hen you are using globalized call routing with local route groups, AAR
implementation is extremely simple: Only a single AAR CSS and AAR group are required and
applied to all phones, regardless of their location.
3-72 ImplementingCisco Unified Communications Manager, Part 2 (CIPT2)v8.0 >2010 Cisco Systems. Inc.
In the other direction, an AAR call from a German phone to a U.S. phone composes a dial
string of 00015115552001. which is the format that is used for inlemalional calls to the United
States. It matches the 0.! route pattern and is sent out using the German gateway.
In summary, this two-site example requires two route patterns in different partitions, two AAR
CSSs. and two AAR groups. In a large, worldwide deployment with lots of different numbering
plans, the configuration of AAR groups can be relatively complex.
There are two sites, one in the United States and the other one in Germany (country codes 1 and
49). At site 1 (country code 1) the access code is 9, at site 2 (country code 49) the access code
is 0. Both countries use 10-digit numbers. There are two route patterns(9.@ for site 1 and 0.!
for site 2). Each route pattern is in a site-specific partition, and the phones use site-specific
CSSs.
Fromthe perspective of AAR, U.S. phones are configured with a 10-digitexternal phone
number mask, and phones at Germany also use national format for the external phone number
mask. U.S. phones are in AAR group U.S., and German phones are in AAR group DE. AAR
prefixes are configured in this way:
• Prefix from AAR group U.S. to AAR group DE: 901149
• Prefix from AAR group DE to AAR group U.S.: 0001
The AAR CSS of U.S. phones has accessto the 9.@ roule pattern; the AAR CSS of German
phones has access to the 0.! route pattern.
When a call from a U.S. phone to a Gennan phone is not admitted because of no available
bandwidth, the extemal phone numbermask of the German phone is merged with the DN of the
phone (in this case, the result is 691253001). Then the prefix 901149configured from AAR
group U.S. to DE is appended, resulting in a call to 901149691253001, which is processed by
the 9.(S) route patternthat refersto the U.S. gateway.
3-70 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 ) 2010 Cisco Systems. Inc.
AAR Characteristics
This subtopic describes the characteristics of AAR.
AAR Characteristics
AAR pro\idesa fallback mechanism for callsthat are denied by locations-based CAC or
RSVP-cnablcd locations-based CAC bv rerouting calls over the PSTN in the event of CAC
failure.
AAR works only for calls that arc placed to internal directory numbers. It docs not apply to
calls that are placed to route patterns or feature patterns such as Meet-Me or Call Park.
Ho\\e\ er. it does work for hunt pilots and computertelephony integration (CTI)ports.These
entities can be configured with an AAR group andan AAR calling search space (CSS).
The alternate number that is used for the PSTN call is composed of the dialed directory
number, a prefix that is configured perAAR source and destination group, and theexternal
phone number mask of the called device.
Alternatively, calls can berouted tovoice mail, oryou can configure an AAR destination mask
for each device that allows an> numberto be used for the rerouted call. The numberthat is
specified at the AAR destination mask isalso known as the Call Forward No Bandwidth
(CFNB) destination.
Note AAR is a fallback mechanism for calls that are denied by locations-based CAC or RSVP-
enabled locations-based CAC. It does not apply to calls that are denied by gateways d le to
exceeding the available or administratively permitted number ofchannels, or to calls tut
have been rejected on trunks (for example, ongatekeeper-controlled H.225 or intercluster
trunks) If suchcalls fail (for whatever reason), fallback mechanisms are provided by route
lists and route groups.
AAR Overview
Headquarters
Branch B
AAR allows calls to be rerouted through the PSTN by using an alternate number when Cisco
Unified Communications Manager blocks a call due to insufficient location bandwidth. With
AAR. the caller does not need to hang up and redial the called party. Without AAR, the user
wouldget a reordertone and the IP phone woulddisplay "Not enoughbandwidth."
AAR applies to centralized call-processing deployments. For instance, ifa telephone in a
company headquarters calls a telephone in branch B and the available bandwidth for the WAN
linkbetween the branches is insufficient (as computed by the locations mechanism), AAR can
reroute the call through the PSTN. The audio path of the call would be IP-based from the
calling phone to its local (headquarters) PSTN gateway, time-division multiplexing (TDM)-
based from thatgateway through the PSTN to the branch B gateway, and IP-based from the
branch B gateway to the destination IP phone.
AAR is transparent to users. Itcan beconfigured so that users dial only theon-net (for
example, four-digit) directory number of the called phone. (Noadditional userinput is required
to reach the destination through an alternate network such as the PSTN.)
In the examplethat is shown here, a call is placed from PhoneA to Phone B, but the locations-
based CAC denies the call due to insufficient bandwidth. Cisco Unified Communications
Managernow automatically composes the required route patternto reach Phone B via the
PSTN and sends the call off-net.
3-68 Implementing Cisco Unified Communications Manager, Part2 (CIPT2) vfl.O 12010 Cisco Systems, Inc.
Afteryou click Save,the changes are displayed in the Location RSVP Settings section of the
window. Only the locations thatarcnotconfigured to use the system default arelisted.
Note You can also enable RSVP within a location. For the currently configured location, Use
System Default is not an option. Youcan choose only No Reservation, Optional (Video
Desired), Mandatory, or Mandatory (VideoDesired)within a location. The defaultfor calls to
own location is No Reservation, and to all other locations, the default is Use System Default.
Note When RSVP-enabled locations are used, it is extremely important that the phones use the
appropriate RSVP agent. Note thattherewill be threecall legs, phone to its RSVP agent;
that RSVP agent to another, remote RSVP agent; and finally, that remoteRSVP agent to its
phone.
How does Cisco Unified Communications Manager determine which RSVP agent is the
RSVP agent to be used by a given phone?The selection of the RSVP agent is based solely
on the MRGLs that are assigned to the phones that attempt to establish a call. Errors in the
MRGL configuration can resultinsuboptimal traffic flows. Therefore, whenyouimplement
RSVP-enabled locations, you must properly assign phones to RSVPagents by using
MRGLs and MRGs. The sample scenarioat the beginning ofthis configuration subtopic
provided all the information that is needed for assigning the RSVP agentstothe phones
Theappropriate configuration is notshown here becausethe configuration ofMRGLs and
MRGswas covered in detail in the Implementing Cisco Unified Communications Manager.
Part 1 (CIPT1) courseand becauseMRGLs and MRGs havealready beendiscussed inthis
course.
©2010 Cisco Syslems. Inc. Bandwidth Management and CAC Implementation 3-67
Step 4: Enable RSVP Between Location Pairs
The figure illustrates how to enable RSVP between pairs of locations.
After configuring the RSVPagentsin Cisco IOS routersand adding them to Cisco Unified
Communications Manager, you need to enable RSVP between one or morepairsof locations.
You performthis task in the Location Configuration window, whichyou access from Cisco
Unified Communications Manager Administration by choosing System > Location. Choose
the location for which RSVP should be enabled for calls to one or more other locations. In the
Location Configuration window underModify Setting(s) to OtherLocations, the currently
configured location is listed. All other locations are also listed.
Choose the location to which RSVP should be used, and then choose the RSVP setting. You
will find the same options thatyou found at the Default interlocation RSVP Policy service
parameter:
• Mandatory (Video Desired): A video call can proceed as anaudio-only call ifa
reservation for the audio stream succeeds but a reservation for the video stream does not
succeed.
Inaddition, there is theoption Use System Default, which applies thevalue of the Default
interlocation RSVP Policy service parameter for calls to the chosen location.
3-66 Implementing CiscoUnified Communications Manager, Part 2 (CIPT2) v8.0 ©2010 Cisco Systems, Inc.
Step 3: Add RSVP Agents to Cisco Unified Communications
Manager
The figure shows how to add an RSVP agent in Cisco Unitied Communications Manager.
^\
Choose media termination point type and device
pool. Enter media termination point name and
description.
Note The configured media termination point name has to matctittie name assignefl lothe
media resource at the Cisco IOS router
After configuring the RSVP agent function at theCisco IOS gateway, youneed to add the
corresponding media resource toCisco Unified Communications Manager. In Cisco Unitied
Communications Manager, choose Media Resources > Media Termination Point and click
Add New.
Inthe Media Termination Point Configuration window, choose the type of the MTP (currently
there is only oneoption. Cisco IOS Enhanced Software Media Termination Point), enter a
name anddescription, and then choose the device pool that should be used.
Note The name of the MTP has to match the name that was configured at the Cisco IOS router
with the associate profile idregister command entered inseep ccm groupidconfiguration
mode. The name is case-sensitive.
Note Because RSVP-enabled locationsallow RSVPto be used between two RSVPagents lat
are betweenthe twoendpoints ofa call, at least two RSVP agents have to be configured in
a cluster to make itwork. Inthe example, these agents are HQ-1 and BR-1 The figure that
is shown with this step is an example that uses the HQ-1 router.
***w
© 2010 Cisco Systems. Inc. Bandwidth Management and CAC Implementation 3-65
Note The bandwidth that is reserved for a call depends on the codec that is used. As with
standard (non-RSVP-enabled) locations, it is 80 kb/s for G.711 and 24 kb/s for G.729.
During the call setup, however, the RSVP agent will always request an additional 16 kb/s,
which is released immediately after the RSVP reservation is successful. Therefore, the
interface bandwidth has to be configured in such a way that it can accommodate the desired
number of calls (considering the codec that will be used) plus the extra 16 kb/s. If, for
example, two G.729 calls are permitted on the interface, 64 kb/s must be configured; for two
2 G.711 calls, 176 kb/s is required.
Note Because RSVP-enabled locations allow RSVP to be used between two RSVP agents that
are between the two endpoints of a call, you need to configure at least two RSVP agents in
a cluster to make it work. In the example, these agents would be HQ-1 and BR-1, The figure
that is shown with this step is an example that uses the HQ-1 router.
3-64 Implementing Cisco Unified Communications Manager. Part 2 (CIPT2) v8.0 ©2010 Cisco Systems, Inc.
Step 2: Configure RSVP Agents in Cisco IOS Software
The figure shows how to configure a Cisco IOS router to enable RSVP-agent functionality.
:p local FastBtheraetO/C
;p ccp 10.1.1.1 identifier 1
;p ccm group 1
Pass-through codec is
SBOciate ecu 1 priority 1
Buociatf profile 1 register hq-1_mtf used, which allows a
CiscolOS Software
MTP to be used.
As with other mediaresources that arc provided by Cisco IOS Software (conference bridges
and transcoders). the configuration starts with global Skinny Client Control Protocol (SCCP)
settings, followed by the Cisco Unified Communications Manager group configuration. In
configuring the media resource itself (which you perform indspfarm profile configuration
mode). \ou use three commands that arc specificto the implementation of a software MI P
RSVP agent:
• codec pass-through: This command specifies that the actual content ofthe RTP stream is
not modified. Mediaresources usually have to interpret and modify the audio stream:
examples are transcoders that change the codec ofthe audio stream, orhardware MTPs that
are used to convertout-of-band signaling to in-hand dual tone multifrequency (DTMF).
The RSVP agent repackages RTP only at Layer 3 and Layer 4. Itterminates the incoming
call leg by dc-encapsulating RTP and then re-encapsulating the identical RTP into a new-
call leg. Because this simple repackaging does not require interpreting and modifying the
audio payload (which isrequired with transcoders or hardware MTPs that are used for
DTMF). the router can perfonn this function in software.
* rsvp: This command specifics that this MTP isused as an RSVP agent that will be used to
set up a call leg to another RSVP agent where RSVP with IntServ over DiffScrv has to be
used.
• maximum sessions software sessions: This command specifies the maximum numberof
sessionsfor the mediaresource. Note that the keyword software has been used. This
keyword indicates that this RSVP agent should nol use digital signal processors (DSPs) but
that itshould perform its function in software. You can use software MTP only when codec
pass-through has been configured.
After setting up the Ml"P RSVP agent, you need toenable RSVP on the WAN interface or
interfaces b\ using the ip rsvp bandwidth bandwidth command. The specified bandwidth
determines how much bandwidth can be reserved by RSVP.
© 2010 Cisco Systems. Inc. Bandwidth Management and CAC Implementation 3-63
Step 1: Configure RSVP Service
Parameters (Cont.)
^Jtfrj :•-•? n-;-j. r.-r^ "i'^sjrti? c*n B* »T,Qt b*tt *HM ^ enor-handling option.
_S«t
or**,* DSCP<OO00O0l v d6T*ultDS
0c**uK DSCP10CQ00D) * d»<Jult 05
3-62 Implementing Cisco Unified Communications Manager, Part2 (CIPT2) v8.0 >2010 Cisco Systems, Inc.
The table shows the interactions of these policy settings.
3-60 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 ©2010 Cisco Systems, Inc.
Step 1: Configure RSVP Service Parameters
Ihis step describes RSVP service parameters and how to configure them.
RVSP Retry Timer: Defines how often (in seconds) the RSVP agent
will retry the reservation if there is a failure.
Mandatory RSVP Mid-Call Error Handle Option: Ifa mid-call failure
occurs, defines whether call becomes best effort or fails (after n
retries).
Mandatory RSVP Mid-Call RetryCounter: Defines the n tries for mid-
call error processing.
— Mandatory (Video Desired): Avideo call can proceed as an audio-only call ifa
reservation for the audio stream succeeds but a reservation for the video stream does
not succeed.
Location HQ Location BR
HQ BRMRGL
10.1.1.1
1
BR RSVP
Cisco Unified
MRG
Communications Manager
l^VtfWt
BR
In the example, there are two sites: headquarters (HQ) and branch (BR). Phones that are located
in the headquarters are in location HQ, and phones that are located at the branch are in location
BR. RSVPagentsexist at each site (HQ-1_MTP is provided by router HQ-1, and BR-1_MTP is
provided by router BR-1). The RSVPagentsare assigned to their respective locations.
Headquarters phones have the MRGL HQ MRGL applied; this MRGL includes the MRG
HQ RSVP_MRG, which includes the HQ-1MTP RSVP agent media resource. Branch phones
have the MRGL BR MRGL applied; this MRGL includes MRG BR RSVP_MRG, which
includes the BR-1_MTP RSVP agent media resource.
Regions (not shown in the figure) are configured in such a way that G.729 has to be used for
calls between headquarters phones and branch phones.
3-58 Implementing CiscoUnified Communications Manager, Part 2 (CIPT2) v8.0 ©2010 Cisco Systems, Inc.
Configuration Procedure for Implementing RSVP-Enabled
Locations-Based CAC
To implement Cisco Unified Communications ManagerRSVP-enabled locations, you will need
to follow these steps.
Because the implementation of Media Resource Groups (MRGs) and MRGLs has been
discussed in detail in the Implementing Cisco Unified Communications Manager, Part I
(CIPT1) course and has been used in earlier lessons of this course, only Steps 1 lo 4 are
discussed in this topic.
• Admission failure
• Bandwidth unavailable
3-56 Implementing CiscoUnified Communications Manager, Part 2 (CIPT2) v8.0 © 2010CiscoSystems, Inc.
How RSVP Works
The figure reviews the basic operation of RSVP.
Resv
Res •; ( Dest 10505050 !!
/ Res* „.„.
Dest 10
,„„30 30 30 . Desl 10 30 30 30 . \NHop 10 SO 60 8'
! Res. ; Dea '.320 20 20 NHop 10 50 50 5o]!\NHop 10 SOSOSrJ
3est 10 10 10 10 \MHap '330 33 30
In Hot. 1020 202\
As shown in the figure, the RSVP-enabled sender (in this case, an RSVPagent)sends a Path
message towardthe RSVP-enabled receiver(again,an RSVPagent in this case) along the path
that requests bandwidth for the call to be set up. The receiver responds with a Resv message
that is routed back along the path. Each RSVP-enabled device cheeks to see if the requested
bandwidth is available and sends the appropriate information in the downstream path toward
the sender.
If no RSVP-enabled device on the path had lo deny the reservation because of insufficient
bandwidth, the reservation was successful: the call was admitted by RSVP CAC.
Here is a more detailed description of the key RSVP messages:
• Path messages (Path): An RSVP Path message is sent by each sender along the unicast or
multicast routesthat are provided by the routing protocol. A Path messageis used to store
the pathstate in each node. The pathstateis used to route Resv messages in thereverse
direction.
) 2010 Cisco Systems. Inc Bandwidth Management and CAC Implementation 3-55
Characteristics of RSVP Agent-to-RSVP Agent Call Leg
This subtopic describes the characteristics of the call leg between two RSVP agents.
The call leg between two RSVP agenLs uses standard RSVP, as implemented in Cisco IOS
routers. The IP network between the RSVP agents is RSVP-enabled. In other words, each
interface is configured with a maximum amount of bandwidth that can be used for RSVP calls.
When not enough bandwidth is available end-to-end (between the two RSVP agents, in this
case). RSVP CAC denies the call,
If RSVP is not enabled on any hop in the path, the appropriate link is ignored by the CAC
algorithm {that is. it is always admitted on this link).
Cisco Unified Communications Manager RSVP agent CAC uses the Integrated Services
(IntServ) and Differentiated Services (DiffServ) models for the RSVP call leg. In other words.
RSVP is used only for CAC (the "control" plane), and not with RSVP-reservable queues for
providing QoS to the streams. Instead, standard low-latency queuing (LLQ) configuration is
required to provision QoS for the voice stream (the "data" plane).
The end-to-end call—that is, the incorporation of all three call legs—is established only after
the RSVP call leg has been admitted. If the RSVP call leg is not admitted, the call fails due to
CAC denial (not enough bandwidth).
3-54 ImplementingCisco Unrfied Communications Manager, Part 2 (CIPT2)v8.0 )2010 Cisco Systems, Inc.
Characteristics of Phone-to-RSVP Agent Call Legs
This subtopic describes the characteristics of the call legs between phones and RSVP agents.
Characteristics of Phone-to-RSVP
Agent Call Legs
Standard locations algorithms apply lo the call leg between an IP phone and its RSVP agent,
which are usually in the same location. If they are in separate locations, standard locations-
based CAC is performed for this call leg (phone to RSVP agent) first. The two RSVP agents
will tr> to set up their call leg by using RSVP only if enough bandwidth is available for the IP
phones to reach their RSVP agents.
An RSVP agent registers with Cisco Unified Communications Manager as a special MTP
device. Cisco Unified Communications Manager uses the Media Resource Group List (MRGL)
of the IP phone to determine which RSVP agent is to be used by which IP phone. The
association of a phone to its RSVP agent does not occur as the result of a search for an RSVP
agent in the same location of the phone. As mentioned earlier, the IP phone and its RSVP agent
can be in separate locations. Only MRGLs arc used to identify the RSVP agent to be used by an
IP phone.
From a design perspective, the RSVPagent that is used by a certain IP phone or group of
phones should be as close as possible to the IP phone or phones. Such a design ensures that
there are optimal pathswhere the phones ideallydo not use the IP WANto accesstheir RSVP
agents. This design also ensuresthat RSVPis used at the IP WAN and that the call legs that do
not use RSVP utilize only LAN infrastructure.
1he RSVP agent supports pass-through codec configuration, which allows any codec to be used
(the codecdocs not have to be known or supported by the RSVP agent). Pass-through codec
configuration includes Secure Real-Time Transport Protocol (SRTP), where the RTP payload is
encrypted.
) 2010 Cisco Systems, Inc Bandwidth Management and CAC Implementation 3-53
Three Call Legs with RSVP-Enabled Locations
When RSVP-enabled locations are used, the end-to-end call is split into three separate call legs.
Location A Location B
— SCCP or SIP
•*• RSVP
** RTP
In the figure, Phonel. which is in Location A, places a call to Phone2, which is in Location B.
The Cisco Unified Communications Manager location configuration specifies that RSVP has to
be used for calls between these two locations.
Cisco Unified Communications Manager instructs the two involved RSVP agents (one in
Location A. and one in Location B) to use RSVP to try to set up the call between each other. If
the call is admitted (that is, if enough bandwidth is available in the network path between these
two devices), the RSVP agents inform Cisco Unified Communications Manager that the RSVP
call leg was successfully set up.
Cisco UnifiedCommunications Managernow tells the phonesto set up their call legs,each to
its respective RSVP agent. If the RSVP call setup between the two RSVP agents is denied,
Cisco Unified Communications Manager considers the call to have failed CAC.
It is important to realize that there are three separate RTP streams: Phonel talks to RSVP
Agentl. RSVP Agentl talks to RSVP Agent2, and RSVP Agent2 talks to Phone2.
RSVP CAC is used between the RSVP agents only.
3-52 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.D >2010 Cisco Systems, Inc.
RSVP-Enabled Locations
"Ihis topic describes RSVP-enabled locations in Cisco Unified Communications Manager.
RSVP-Enabled Locations
Characteristics
• Link failures: If one link in the IP network goes down and packets are routed on different
paths. RSVP is aware of the change and considers the bandwidth thatis now available at
the path that is actually routed.
• Backup links: Ifbackup links are added after link failures, orif bandwidth ondemand is
used to add dial-on-demand circuits. RSVP again is fully awareof the routing path that is
currently used and thebandwidth that is available on each link along thatpath.
• Load-share paths: If load sharing is used. RSVP is aware of the overall bandwidth thatis
provided by multiple load-sharing links.
Using RSVP for CAC allows admitting ordenying calls that are based onactual
oversubscriptions. The result is always based onthe currently available bandwidth and
interfaces, not on a logical configuration that ignores the physical topology.
Cisco Unified
Communications Product Tr»»: tltt*
OtvHe Proton 1 SCCP Location is
Manager indirectly applied
Administration: BBQi[tr*(*n ur-tnown via device pool
Device > Phone (each device
pool Is
H"*"™'1 cm* —= T
location).
Itisr-f &L*ron Tarrtpl&U*
j If location is
1 selectedhere.
J location of
device pool is
1 ignored
N„„*m0w«om.u*.s
Locations are a mandatory setting in a device pool, and you must assign a device pool to each
device. Therefore, a device always has a location that is assigned indirectly through its device
pool. If a device uses a different location from the one specified in its device pool, that location
can be chosen at the device itself. A location that is assigned at the device levelhas higher
priority than the location of the device pool.
3-50 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 12010 Cisco Systems, Inc.
Step 1: Configure Locations
The figure shows the configuration of the locations in Cisco Unified Communications
Manager.
Location HQ 96 kb/s
There are three sites: the headquarters and two branches. Each site has its own location (IIQ,
BR I. and BR2). The physical topology is a hub-and-spoke topology (headquarters is the hub).
The link between branch I and the headquarters should not carry more than one G.729 call, and
the link between branch 2 and the headquarters should not carry more than three G.729 calls.
The next two subtopics describe how to implement locations-based CAC for this scenario.
3-48 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8,D >2010Cisco Systems, Inc.
Configuration Procedure for Implementing Locations-Based
CAC
To implement Cisco Unified Communications Manager locations-based CAC. you need to
follow two steps.
Note To know how much bandwidth you need to calculate per call, you should design and
configure regions before implementing locations
© 2010 Cisco Systems, inc Bandwidth Management and CAC Implementation 3-47
Locations: Full-Mesh Topology
The figure shows a full mesh topology with locations-based CAC.
This example is based on the previous example, but a direct IP WAN link has been added
between BRl and BR2. The idea is that one G.729 call is allowed on the WAN link from BRl
toward the headquarters, one G.729 call is allowed on the WAN link between BRl and BR2,
and three G.729 calls are allowed on the WAN link from BR2 toward the headquarters.
Such a scenario reveals issues that arise when locations-based CAC is used in topologies other
than hub-and-spoke topologies. To allow the additional G.729 call that is permitted on the
WAN link between BRl and BR2, the bandwidth limit of these two locations has been
increased by 24 kb/s. Increasing the bandwidth, however, can lead to these undesirable
situations:
• Two G.729 calls from BRl to HQ: Because the BRl location now has a limit of 48 kb/s.
it allows two G.729 calls. Location bandwidth limits are not configured per destination; any
call coming into or going out of a location is considered, regardless of the other location
that is involved in the call. Therefore, there is no way to divide the available 48 kb/s into
one call toward the HQ and one call to BR2.
• Kour G.729 calls from BR2 to HQ: The same problem occurs with the BR2 location: The
additional bandwidth that was added to accommodate the desired call toward BRl can be
used toward I IQ. occupying that link with one more call than intended.
Note The problems that are described here are caused by the fact that the bandwidth limit is
configured per location, regardless of the other location (where the call goes or comes from)
3-46 Implementing Cisco Unitied Communications Manager, Part 2 (CIPT2) v8.0 12010 Cisco Systems, Inc.
Locations: Hub-and-Spoke Topology
"fhe figure shows a hub-and-spoke Cisco Unified Communications Manager topology with
locations-based CAC.
lxG.729 I \ 3xG729
IPWANV
As shown in the figure, there are three sites: the headquarters (HQ) and two branches (BRl and
BR2). There is no direct connection between the branches; all traffic goes by way of the
headquarters.
Ihis scenario is ideal for locations-based CAC If the intention is lo allow only one G.729 call
on the link between BRl and HQ and three G.729 calls on the link between BR2 and HQ. this
location configuration would suit these needs:
• Location HQ: Unlimited
This coniiguration ensures that no more than oneG.729 call will be sentoverthe IP WAN
toward location BRl and that no more than three G.729 calls will be sent over the IP WAN
toward location BR2.
Note The configuration also allows one G.729 call betweenBR1 and BR2. Because the
configured bandwidth limit does notconsider the destination location, the 24-kb/s limit of
BR1 allows any call to go out (orcome in) regardlessof where itgoes (orwhere itcomes
from) The headquarters limit is not affected at all bysucha call. Only locations BR1 and
BR2 will subtract 24 kb/s from their limits. Because locations-based CAC does not provide
topology awareness, Cisco Unified Communications Manager is not even aware that thecall
physically flows through the headquarters. ^^
Locations Characteristics
Each device has one location assigned. The assignment can be direct or via a device pool. If
both types of assignment are used, the device configuration has higher priority.
You limit calls by permitting a certain bandwidth for all calls coming into and going out of a
location. Cisco Unified Communications Manager calculates the actual audio codec bandwidth
plus IP overhead (assuming a packetization period of 20 ms). This means that each G.711 call
reduces the bandwidth that is configured for a location by 80 kb/s, while a G.729 call reduces
the available bandwidth by 24 kb/s.
Note Calls withina location do not decrease the bandwidth limit; they are unlimited. Only calls that
go out of a location or that are received from outside the location are considered by the
locations-based CAC algorithm.
The bandwidth limitsthat are configured at the location of the originating device (the source
location) as well as at the location of the terminating device (the destination location) are
checked individually. Unlike with regionconfiguration, where the maximum permittedcodec is
configured perpairofregions, the bandwidth limitof a location applies to all (both placed and
received) interlocation calls. If the bandwidth limit of the source or of the destination location
(or of both) is exceeded, the call is not admitted. Locations provideCAC for calls within
clusters; however, because locations can alsobe configured for gateways and trunks, locations
do allow some control for calls leaving the cluster.
Locations-based CAC in Cisco Unified Communications Manager is completely unaware of the
topology of the network. It is a purely logical assignment and does not reflect the actual
topology or the actual bandwidth available.
3-44 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 ) 2010 Cisco Systems, Inc.
CAC in Cisco Unified Communications Manager
Cisco Unified Communications Manager supports various CAC methods.
RSVP-enabled locations
In centralized call-processing deployments, you can use standard locations and Resource
Reservation Protocol (RSVP)-enabled locations lo provide CAC within a Cisco Unified
Communications Manager cluster. If a call is not admitted by one of these two CAC methods
due to bandwidth limitations, you can use AARto reroutethe call over the PSTN (off-net)
instead of denying the call. AAR provides a service like PSTN backup, except thatthe reason
for call backup is not that the call failedon the on-ncl path, but that there is no available
bandwidth from a CAC point of view.
In distributed call-processing environments, you canuse H.323 gatekeeper CAC with H.323
trunks (gatekeeper-controlled intercluster trunks and H.225 trunks). If Session Initiation
Protocol (SIP)trunks, you can use SIP Preconditions, which allows RSVP-based CAC.
If calls are not admitted by the H.323 gatekeeper, standardbackup functionalitv of route lists
and route groups is applied, for example, to route calls thathave not been admitted by the
gatekeeper to be sent over the trunk, you can configure one or more PSI'N gateways inanother
(lower-priority) route group of the same route list. In this way, the gatekeeper-controlled trunk
ispreferred over the PSTN aslong ascalls are admitted; after admission isrejected, calls are
sentoverthe PSTN. Thesame principle applies to callsthat arc placed through SIP trunks that
are configured for SIP Preconditions.
CAC limits the number of calls between certain parts of the network in order to avoid
bandwidth oversubscription with too many voice calls. QoS is not able to achieve this result
because QoS provides only the means to prioritize Voice over Data traffic. QoS does not avoid
the situation in which too many (prioritized) voice streams are sent over the network.
If oversubscription occurs, any packets of any voice stream can be affected, not just packets of
the particular call or calls that exceed the bandwidth limit. The result in this case is packet
delays and packet drops of all voice calls, and hence oversubscription degrades the quality of
all voice calls.
Therefore, in order to ensure good voice quality, you need to use CAC to limit the number of
voice calls.
3-42 ImplementingCisco Unified Communications Manager, Part 2 (CIPT2)v8.0 >2010 Cisco Systems. Inc.
Lesson 2
Implementing CAC
Overview
Implementing multisite IP telephony deployments over an IP WAN requires additional
planning to ensure the quality and availability of voice calls.
When an IP WAN connects multiple sites in a Cisco Unified Communications deployment.
quality of sen. ice (QoS) hasto be implemented in orderto prioritise voice packets overdata
packets. Ilow ever, to avoid an oversubscription that is caused by too many voice calls, a
mechanism is necessary to limit the numberof calls that are allowed at the sametime between
certain locations. Call Admission Control (CAC) is the mechanism that ensures that voice calls
do not oversubscribe the IP WAN bandwidth and thus impact voice quality.
This lesson describes how to implement CAC mechanisms that are provided by Cisco Unified
Communications Manager, andexplains how automated alternate routing (AAR) can be used in
some scenarios lo reroute calls that were denied by CAC over the public switched telephone
network (PSTN).
Objectives
Upon completing this lesson, you will be able to describe and configure CAC mechanisms and
AAR in Cisco Unified Communications Manager and in gatekeepers. Ihis ability includes
being able to meet these objectives:
,^M u Describe the CAC options that are provided by Cisco Unified Communications Manager
• Implement locations-based CAC in Cisco Unified Communications Manager
• Implement RSVP-enabled locations-based CAC in Cisco Unified Communications
*•* Manager
• Implement AAR inorder to reroute intracluster calls over the PSI'N if notenough
bandwidth is available for an on-net call
• Implement SIP Preconditions onSIP trunks inCisco Unified Communications Manager
• Implement 11.323 gatckceper-based CAC in Cisco Unified Communications Manager
3-40 Implementing Cisco Unified Communications Manager, Part2 (CIPT2) v8.0 ©2010 Cisco Systems, Inc.
Summary
This topic summarizes the key points that were discussed in this lesson.
Summary
References
for additional infonnation. refer to these resources:
• Cisco Svstems. Inc. Cisco Unified Communications System 8.x SRND. April 2010.
litip:.,',www.ci,bCo.com/en.llS'/docs/\oicc ip eomm/cucm/srnd/8\/ue8\.htm)
• Cisco S\ stems. Inc. Cisco Unified Communications Manager Administration Guide
Release 8.0(1). February' 2010.
iilip:';www.cisc(XC<)m;cn;US^docs^(iiec_ip_coniin/cucni/admin/8_0_l/ecnicfg''1iccm-801-
cm.him]
• Cisco Systems. Inc. Cisco Unified SRST System Administrator Guide. December 2007.
hup:.' www,cisco.eoni/en'US/pji'lner'docs/voice ip conim/cusrst/admiii/srst/coiifiguratioii.'
sjuide.-'srstsa.html
ip multicast-routing
interface FastEthernetO/0
description HO-Voice-Servers
ip address 10.1.1.101 355.255.255.0
ip pim sparse-dense-mode
interface FastEthsrnatO/0
description HQ-Phones
ip address 10.1.2.101 255.255.255.0
ip pim sparse-dense-mode
interface SerialO/l
If no other multicast applications are used over the IP WAN, the simplest way of preventing the
multicast MOH packets from being sent to the WAN is to disable multicast routing at the WAN
interface.
3-38 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 ) 2010 Cisco Systems, Inc.
Step 4b: Use IP ACL at IP WAN Router Interface
The figure shows how you can use an IP ACL to configure a Cisco IOS router to drop multicast
MOI I packets.
ip multicast-routing
interface FastEthernetO/1
description HO-Phones
ip address 10,1.2.101 255.255.255.0
ip pim sparse-dense-mode
interface SeriolO/1
description IP MAN
ip address 10.1.4.101 255.255.255.0 [" PACLAppliedtoWAN
ip access-group drop-moti out ^___ 1
nterface
ip pim sparse-dense-mode i
The ACL matches the MOH group address and port numbers that are used by the MOII server
for the MOH RTP and RTCT packets. The ACL is applied to the IP WAN interface in the
outgoing direction and therefore doesnot allow multicast MOH packets to be sentout on the IP
WAN.
Note As stated earlier in this lesson, the multicast address and port range that must be filtered
depend on several parameters, such as the audiosource number, the enabledcodecs, and
the increment method.
(Utdroi*
&*u»U&t4U»Brt 16SW
*""*" numbn null)
ta™.»tau ^ hnHfan*,. ' if Mai it
All MOH audio sources that have been configured for multicasting are listed in the Selected
Multicast Audio Sources section of the MOH Server Configuration screen. You can set the
Max Hops value for each audio source; the default is 2. This parameter sets the TTL value in
the IP header of themulticast MOH RTP packets to the specified value. TTLin an IP packet
indicates the maximum numberof routersthat an audio source is allowedto cross. If Max Hops
is set to 1. the multicast MOH RTP packets remain in the subnet of the multicast MOH server.
Whenyou use multicast MOHfrom branch router flash, you can set Max Hopsto a value that
is lower than the actual hop count from the MOI 1server toward the WAN interface of the main
site router. This value, however, might conflict with the needs within the main site when IP
phonenetworkshave the same or a higher distance—that is, a higherhop count—to the MOH
serverthan the WAN network. In such a case, one of the other possible methodsof preventing
the multicast MOH packets that aregenerated by the MOH server have to be used. Theyare
shown on the following pages.
3-36 Implementing CiscoUnified Communications Manager, Part 2 (C1PT2) v8.0 >2010 Cisco Systems, Inc.
Step 3: Enable Multicast MOH from Branch Router Flash at the
Branch Router
The figure shows how to configure a Cisco IOS routerfor multicast MOI I from branch roi ter
flash.
call-manager-fallback
mai-epbones 1
nax-dn 1
ip source-address 10.1.5.102
nob mob-file.au
multicast mob 239.1.1.1 port 16364
interface FastEtharnatO/0
description BF-Phones
ip address 10.1.5.102 255.255.255.0
interface SerlalO/1
description IP WXN
ip address 10.1.4.102 255.255.255.0
Multicast MOH from branch router flash is part of the SRST feature. Therefore, SRST must
already beconfigured before youcan enable multicast MOII from branch router flash.
Note SRSTconfiguration options are discussed inthe module "Centralized Call -Processing
Redundancy Implementation."
Rased on anexisting SRST configuration, you need only two commands to enable multicast
MOH from branch router Hash:
• moh file-name: fhis command specifies the MOH audio source file. The specified file has
to be stored in flash memor> of the SRST gateway.
• multicast moh multicast-group-address portport: This command specifies themulticast
address and port that are used for the multicast MOH packets. The specified address and
port have toexactly match the values that have been configured at the MOH server in
Step 2b.
Note TheSRST gateway will permanently streamMOH, regardless ofan IPWAN failure or IP
phones being registered with the SRSTgateway. ^^
You can configure an additional five MOH streams using MOH group configuration. Refer to
the module "Centralized Call-Processing Redundancy Implementation" for more information
about MOII group configuration.
Multicast MOH only works if the multicast enabled MOH server is assigned to a multicast
enabled MRG. This MRG will be configured to be a member of an MRGL. The MRGL will
then be associated with devices such as phones.
%m*
3-34 ImplemenlingCisco Unified Communications Manager, Part 2 (CIPT2)v8.0 >2010 Cisco Systems, Inc.
«*»
Step 2b: Configure Multicast MOH in Cisco Unified
Communications Manager
After allowing multicast MOH on audio sources, you must enable the MOH server for
multicast MOH. as shown in the figure.
r^ttmtlkijvltvtr InlwnilMn- D J
Enable multicast MOH on the
•iaMtMHfcWiiliwi.in rmm itniif MOH server.
'.ferine**'
fhe figure shows how to enable multicastMOH on a MOH server: In the Multicast Audio
Source Infonnation section of the MOH server configuration screen, check the finable
Multicast Audio Sources on this MOH Server check box. 'fhe Base Multicast IP Address, Jase
Multicast Port Number, and Increment Multicast On parameters are automatically populated
when you enable multicast MOH on the server. You can modify these values as desired.
Note To avoid network saturation in firewall situations, it is recommended that you choose to
increment multicast MOH on the IP address instead of on the port number. Choosing this
option means that each multicastaudio source will have a unique IP address, and helps to
avoid network saturation If multiplecodecs are enabled for the MOH server, additional IP
addresses will be in use (one per codec and per audio source).
>2010 Cisco Systems, Inc. Bandwidth Management and CAC Implementation 3-33
Step 2a: Configure MOH Audio Sources for Multicast MOH
To enable multicast MOH. you first have to allow multicast MOH on MOH audio sources, as
shown in the figure.
-Nuikh W d l r m r h A t b V f l l**W1**Oan-
Mm'
|l 'u»Mc«»i| |
("0-T*.( Tr#ftjl»4tfift Coft*l*« HemMi (M^jksJ.Ni™ ntouhrd j
DilfcSjKI-6
LO-Ctf •T.-i*
Btf«
11 )I»H J 1 i
\
MOH audio sources do not allow multicast
tan pi* Audi iSotii ca Lul*- t*a y MOH by default.
MOH audio sources and fixed MOH audio
irra^'udi^i^t* (?28.i««v
"
sources (if used) must be enabled for
multicast MOH.
Check the Allow Multicasting check box for each MOH audio that is allowed to be sent as a
multicast stream. This instruction applies to MOH audio sources and to fixed MOH audio
sources.
Note More information about configuring the MOH server and MOH audio sources is provided in
the ImplementingCisco Unified Communications Manager, Part 1 (CIPT1) course.
3-32 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) vB.O © 2010 Cisco Systems, Inc.
Step 1: Enable Multicast Routing on Cisco IOS Routers
The figure shows how to enable multicast routing on Cisco IOS routers at the main site an<: at
the remote site.
ip multicast-routing ~ —
Enable multicast
interface FaatEthernetQ/O
routing on router.
description HQ-Voice-Servers
ip address 10.1.1.101 255.255.255.0
ip pin sparse-dense-mode
interface FaatEtnernetO/1
description HQ-Phonea
ip address 10.1.3.101 255.255.255.0
ip pim sparse-dense-mode Enable multicast
routing on interfaces.
interface SerialO/l
description IP WAN
ip address 10.1.5.101 255.255.255.0
ip pim sparse-dense-mode
You use two commands to enable multicast routing in the network so that multicast MOH
streams can he sent:
Note The configuration that is shown in the example enables multicast routing in the whole
network. When multicast MOH from branch router flash is used, multicast streams will not be
sent to the IP WAN They can be blocked based on the maximum hops parameter (TTL field
in the IP header) or by IP ACLs. You can also block multicast streams by disabling multicast
routing on the interface, butonlyifno other multicast routing applications are required inthe
network.
In the example, the maximum hops parametercannot be used because the HQ-Phones
network and the IP WAN network have the same distance to the HQ-Voice-Servers network
To allow multicast MOH to be sent to the HQ phones, a maximum hop value of 2 is required.
This value, however, will allowthe multicast MOH packets to be sent out on the WAN
interface. Therefore, IP ACLs have to be used, or multicast routing has to be disabled at the
WAN interface (if multicast routing is not required by otherapplications).
The configuration procedure describes the implementation of multicast MOH from branch
router flash by first enabling multicast MOH (steps 1 and 2). Once this works as desired, the
configuration is modified so that the multicastMOH streamis generated locallyat the branch
router (Step 3) and the multicast MOH stream that is generated by the MOH server is prevented
from being sent to the IP WAN.
WhenenablingmulticastMOH at the MOHserver,make sure that you set the maximum hop
value of the multicast-enabled MOH audiosource(s)to a high enough value to allow the
multicast MOH packets to be sent all the way to the remote phones.
When choosing option 4a to preventing the multicast MOH stream of the MOH server from
being sent to the IP WAN.you have to use a low enough value to ensure that the multicast
MOH packets generated by the MOH server do not reach the IP WAN.
Note All IP phones must be able to access to the main site Cisco Unified Communications
Manager MOH server from their MRGL. This access is required as soon as multicast MOH
is configured, whether multicast MOH from branch router flash is used. If the remote site IP
phones do not have access to the Cisco Unified Communications Manager MOH server
from theirMRGL, CiscoUnified Communications Manager cannot instruct the IP phones to
join the multicast group and will make the phone use tone on hold instead of MOH.
Furthermore, you need to check the Use Multicast for MOH Audio check box at the MRG
that includes the multicast-enabled MOH server (see Step 2c).
Finally, make sure that the G.711 codec is used between the MOH server and the branch
phones, because SRST multicast MOH supports only G.711.
3-30 Implementing CiscoUnified Communications Manager, Part 2 (CIPT2) v8.0 ©2010 Cisco Systems, Inc.
• Disable multicast routine at the IP WAN interface: By disabling multicast routing at the
IP WAN interface, multicast packets are not routed out on that interface.
At the branch router, the multicast MOH stream is sent out on the interface that is specified in
the ip source address command in call-manager-fallback coniiguration mode(or in telephony-
server configuration mode, when Cisco Unified Communications Manager Express in SRST
mode is used). Therefore, the multicast MOII stream that is generated at the branch router does
not have to be blocked at the branch router WAN interface.
©2010 Cisco Systems, Inc Bandwidth Management and CAC Implementation 3-29
Example: Implementing Multicast (VIOH
from Branch Router Flash (Cont.)
Cisco Unified
Communications
Manager MOH
Configuration
DA 239 1 1 1
DP. 16384
(a) Max Hops
(TTLV1
Main Site
ill-manager-fallback
i-epbones 1
(b) ip accaaa -liat sirtended drop mob majt-dn 1
deny udp any ho 3t 239.1 1.1 i ange ip aouroa-address 10.1.5.102
16384 16385
moh moh-fila.an
per•Bit ip any any
multicast moh 239.1.1.1
inte rface serial 0/0 port 16394
ip access -group drop-moh out
Now the multicast MOI1stream that is sent toward the remote site needs to be blocked, and
multicast MOH from branch router flash needs to be implemented at the remote site.
Therefore, the SRST configuration of the remote site router is extended to include multicast
MOH. The SRST configuration uses the same multicast IP address and portthat are configured
at the Cisco Unified Communications Manager MOH server that is located at the main site.
To stopmulticast MOH generated by themain siteCisco Unified Communications Manager
MOH server from beingsentoverthe IP WAN, you canchoose oneof three options:
• Set Time to Live (TTL) to a low enough value at the Cisco Unified Communications
Manager MOH server: If the TTL value in the IP headerof the generated multicastMOH
packets is set to a low enough value, the packets will not be routed out to the IP WAN.
However, if the IP WAN link is one hop away from the Cisco Unified Communications
Manager MOII server, andif the main site phones are alsoone hop away from the server.
this method cannot beused, because themain site IPphones would also be affected by the
dropped packets. In the current example, TTL is set to 1, and it is assumed that the IP
phones are inthe same VLAN, like the Cisco Unified Communications Manager MOI I
server.
• Filter the packets by an IP access control list (ACL): At the main site router, an ACL
can be configured that drops the multicast MOH packets at the IP WAN interface.
Note Make sure thatyou verify the actually used multicast IPaddresses and ports. Asdescribed
earlier, itdepends on the base address and port configuration, the method that is used to
increment the base number {on IP address or port), the codecs that are enabledfor MOH,
and the audio sources that are multicast-enabled.
3-28 Implementing Cisco Unified Communications Manager, Part2 (CIPT2) v8.0 ©2010 Cisco Systems, Inc.
Example: Implementing Multicast MOH from Branch Router
Flash
The figure shows a sample scenario for implementing multicast MOH from branch router flash.
Cisco Unified
Communicalions
Manager MOH
Wain Site
WAN
In the example, a MOH server is located at themain site. It is configured for multicast MOH.
Multicast routing has been enabled in the whole network—including inthe IP WAN link to the
remote site.
The main siterouter, however, should no longer route multicast MOH to the remote site. The
remote site SRSI gateway should instead generate multicast MOII streams to the phones that
are located at the remote site.
Cisco Unified Communications Manager is not aware that the multicast packets that are
generated by the MOH server atthe main site are filtered onthe IP WAN interface and then are
locally generated by the remote site SRST gateway. Therefore, Cisco Unified Communications
Manager will instruct the IP phones that are located at the remote site lojoin the multicast
group IP address that isconfigured at the Cisco Unified Communications Manager MOH
server. To allow the phones to receive MOH for the multicast group IP address that they join,
you must configure the SRST gateway touse the same multicast address and port that isused
bv the Cisco Unified Communications Manager MOH server that is located at the main site.
Note The output that is shown does not match the example in the figure. It is used only to
illustrate which information you will find in the trace output. The number after the
KickStartMultiCastStream identifies the audio source. For each enabled codec, you will find
information about the used multicast IP address and port. The NID (node ID) shows the IP
address of the MOH server. In this example, only G.711 mu-law and G.711 a-law codecs
are enabled. Onlyone audio source (audio source 1) is multicast-enabled. There is a single
MOH server at 10.1.1 1
Tip It is importantto know the used multicastIP addresses and ports when you choose the
option to prevent multicast traffic from entering the IP WAN by access lists.
3-26 Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 ©2010 Cisco Systems, Inc.
Multicast MOH: Address and Port Increment Example
The table shows an example of IP address and port increments for multicast MOH.
Audio Source 2 No
Audio Source 4 No
Audio Source 5 No
The base multicast group iseonligured for IPaddress 239.1.1.1 and port 16384. G.711 a-law
and G.729 codecs are enabled: audio sources 1. 3. and 6 are multicast-enabled.
The figure slums the IP addresses orports that are used for the actual multicast MOII streams.
The next table shows how these numbers were derived.
Note The yellow highlighted numbers mthetable present thevalues thatare used in thefigure.
The gray highlighted and bold numbers present theIP address and port number that are
actually used in this example.
For each audio source, four streams are considered (one per
codec: G.711 mu-law, G.711 a-law, G.729, and wideband) for the
increment.
Because a single MOH server can stream multiple multicast MOH files, you have to specify an
initial multicast address and portthat is used forthe firststream. In addition, you have to
choose whether to increment the IP address or port on additional streams. It is recommended
thatyou increment on IP addresses instead of on ports. If there aremultiple MOH servers
within a network, you have to make sure that they do not use overlapping multicastIP
addresses and ports for their streams.
Foreach audio source, fourstreams are considered for the increment—one per codec: G.711
mu-law, G.711 a-law, G.729, and wideband. Thisprinciple always applies, regardless of which
MOH codecs have been enabled in the CiscoIP Voice Media Streaming Application service.
When you are incrementing on IP addresses, each stream consumes one IP address. In other
words, eachaudio source requires fourIP addresses. When incrementing on ports, you have to
considerthe Real-Time Transport Control Protocol (RTCP). For each audio stream,two
separate RTP ports are reserved: one forthe actual audiotransmission andone for (the
optional) RTCP. Therefore, when you areincrementing multicast MOH on ports, each stream
consumes two ports. You have to calculate eight port numbers peraudio source (two ports per
codec).
Audio sources that are not enabled for multicast MOH should nevertheless be considered for
the increment ofaddresses orports. Audio source 1,which starts with the configured base
address and port, requires four IPaddresses oreight ports. The same principle applies toeach
consecutive audio source (audio source 2, audio source 3, and so on), regardless of whether
these audio sources are multicast-enabled.
3-24 Implementing CiscoUnified Communications Manager, Part2 (CIPT2) v8.0 © 2010 Cisco Systems, Inc.
Multicast MOH from Branch Router Flash: Region
Considerations
Ihis subtopic discusses how to configure codecs when you are using multicast MOH from
branch router flash.
When multicast MOH is used. IP phones andCisco Unified Communications Manager are not
aware that the IP phones listen to locally generated MOH streams. From a signaling
perspective, the IP phone isinstructed to listen to acertain multicast stream, and the local
SRSI gateway has to generate amulticast MOH stream by using identical settings, such as
destination address (multicast group), destination port, codec, and packetization period.
Multicast MOH in SRST gateways and Cisco Unified Communications Manager support only
the G.711 codec, 'fherefore. G.711 must also be configured between the Cisco Unified
Communications Manager MOH server and the branch IPphones. IfCisco Unified
Communications Manager signals a codec other than G.711 tothe IP phone, the IP phone could
not play the locally generated MOH stream because ofa codec mismatch (the signaling would
be G.729. but the received RTP stream would be G.711).
Toensure that Cisco Unified Communications Manager sends signaling messages to the phone
and instructs it to listen to a G.711 stream,configure regions in this way:
• Put the Cisco Unified Communications Manager MOH server or servers into a dedicated
region (for example. MOH).
• Putall branch devices intoa site-specific region (forexample. Branch-1).
• Allow G.711 between regions MOH and Branch-1.
• Make sure that region Branch-1 islimited to G.729 for calls loand from all other regions.
3-22 Implementing Cisco Unified Communications Manager, Part 2(CIPT2) w8.0 ©2010 Cisco Systems, Inc.
Multicast MOH from Branch Router Flash
Implementation
1his topic describes how to implement the multicast MOH from branch router flash feature.
Multicast MOH from branch router flash is a feature that allows multicast MOI I streams to be
generated b\ gateways that are located at remote sites instead ofbeing streamed from the main
site to the remote site over the IP WAN.
Cisco Unified Communications Manager is configured for standard multicastMOH. Neither
Cisco IInilied Communications Manager northe phones that are located at theremote siteare
aware that the stream generated at the centra! site is replaced by a locally generated stream. The
multicast MOH stream that is generated by the centrally located MOH server is prevented from
traversing over IPWAN. and the remote site router generates a stream that has the same
attributes (codec, multicast address, and port).
As mentioned earlier, multicast MOII from branch router flash is based on multicast MOH. so
\ou must configure Cisco Unified Communications Manager to use multicast MOH instead of
unicast MOH. This configuration is recommended anyway in order to reduce load at the MOH
server by multicasting one stream that can be received by all devices, instead ofstreaming
MOH indi\idually for each endpoint in separate RIP sessions.
To generate a multicast MOI Istream at the remote site, you use features ofSurvivable Remote
Site Telephom (SRST) orCisco Unified Communications Manager Express, fherefore. the
remote site router thatwill generate the multicast MOI I stream forthedevices that are located
at the remote site has to beconfigured for SRST or Cisco Unified Communications Manager
Express. SRSI" does not have to be active (there is no need for a fallback scenario), because an
SRST gatewaj that is configured for multicast MOH streams MOH all the time, regardless of
its state (standby mode or SRST mode). The same principle applies toCisco Unified
Communications Manager Express: Only multicast MOH has tobe enabled, no further features
have to be enabled,and no phones have to be registered.
Tip The name that is specified in the Cisco IOS device must match the name in the Cisco
Unified Communications Manager exactly; the names are case-sensitive.
Note When a Cisco IOS Enhanced Media Termination Point is being configured, any name can
be configured with the associate profile command. When a Cisco IOS conference bridge is
being configured, the name cannot be configured; it is MTP(AMC), where (MAC) is the MAC
address of the interface that was specified at the seep local command.
• dspfarm profile: To enter DSP farm profile configuration mode and define a profile for
DSP farm services, use the dspfarm profile command in global configuration mode.
• codec (dsp): To specify call density and codec complexity that is based on a particular
codec standard, use the codec command in DSP interface DSP farm configuration mode.
• maximum sessions (DSP farm profile): To specify the maximum number of sessions that
are supported by the profile, use the maximum sessions command in DSP farm profile
configuration mode.
• associate application seep: To associate SCCP to the DSP farm profile, use the associate
application seep command in DSP farm profile configuration mode.
• no shutdown: If you fail to use the no shut command for the DSP farm profile, it will be
displayed in the gateway but will fail to operate.
To verify the Cisco IOS media resource configuration, use fhese show commands:
• show seep: To check whether the Cisco IOS router successfully established a TCP
connection with the configured Cisco UnifiedCommunications Managersystem or systems
in order to exchange SCCP signaling messages, use the show seep command.
• show seep ecm group [group-number]: To see which media resources are registered with
the Cisco Unified Communications Managersystem or systemsthat are configured in the
specified group, use the show seep ccm group / command.
• show dspfarm profile [group-number]: To see the status of the media resource of the
specified profile at the Cisco IOS router, use the show dspfarm profile / command.
3-20 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 ©2010 Cisco Systems, Inc.
Step 2: Configure Transcoder Resource in Cisco IOS Software
The figure shows how to configure Cisco IOS Software to provide transcoding resources to
Cisco Unified Communications Manager.
In theexample that is shown in the figure, a Cisco IOS Enhanced Media Termination Point
type transcoder is configured:
• dspfarm (DSP farm): To enable DSP farm service, use the dspfarm command in global
configuration mode. The DSP farm service is disabled by default.
• dsp sen ices dspfarm: To enable DSP farm services for a particular voice network
module, usethe dsp servicesdspfarm command in interface configuration mode.
• seep local: To use Skinny Client Control Protocol (SCCP) to select the local interface that
is used to register the media resources with Cisco Unified Communications Manager, enter
the seep local command in globalconfiguration mode.
• seep ccm: To use SCCP to add a Cisco Unified Communications Manager server tothe list
of available serversand set variousparameters—including IP addressor Domain Name
System (DNS) name, port number, and version number—use the seep ccm command in
global configuration mode.
• seep: To enable the SCCP protocol and its related applications (for example, transcoding
and conferencing), use theseepcommand inglobal configuration mode.
• seep ccm group: To create a Cisco Unified Communicalions Manager group and enter
SCCPCisco Unified Communications Managerconfiguration mode, use the seep ccm
group command in global configuration mode.
• associate ccm: To associate a Cisco UnifiedCommunications Manager with a Cisco
Unified Communications Manager group and establish itspnority within the group, use the
associate ccm command in SCCPCisco Unified Communications Managerconfiguration
mode.
Navigate to Media Resources > Conference Bridge and click Add New. The Transcoder
Configuration window opens. Choose the type of Cisco transcoder media resource from these
options:
• Cisco IOS Enhanced Media Termination Point
Note The type depends on the hardware that is used. For example, NM-HDV would require Cisco
IOS Media Termination Point to be selected while newer DSP hardware such as NM-HDV2
is configured as Cisco IOS Enhanced Media Termination Point.
Choosethe type of the Ciscotranscoder mediaresource, enter a device name and a description
for the transcoding resource, and then choose a device pool.
The device name has to match the name that is entered at the Cisco IOS router thai provides the
mediaresource. The name is case-sensitive. If the transcoding resource is provided by Cisco
IOS Enhanced Media Termination Point hardware, you can freely choose the name. In all other
cases,the name is MTPfollowed by ihe MAC addressof the interface that is configured to be
used for registering the media resource with Cisco Unified Communications Manager.
3-18 Implementing Cisco Unrfied Communications Manager, Part 2 (CIPT2)v8.0 >2010 Cisco Systems, Inc.
Configuration Procedure for Implementing Transcoders
"fhis subtopic presents the procedure for implementing transcoders.
Note Only the first two steps of the procedure are presented on the following pages, because the
last three steps—the configuration of MRGs, the configuration of MRGLs, and the
assignmentof MRGLs to devices—are discussed inthe Implementing Cisco Unified
Communications Manager, Part 1 (CIPT1) course.
©2010 Cisco Systems. Inc Bandwidth Management and CAC Implementation 3-17
Example: Implementing a Transcoder at
the Main Site (Cont.)
Region: HQ Region: BR
Region: HQ
The figure illustrates how the solution described earlier in this topic is implemented in Cisco
Unified Communications Manager.
All headquarters devices (phones, voice-mail system, software conference bridge, and the
transcoder) are in region HQ. Remote site phones are in region BR.
Cisco Unified Communications Manager region configuration allows G.711 to be used within
region HO and within region BR. Calls between regions HQ and BR are limited to G.729.
When a call is placed from a remote site phone to the voice-mail system, Cisco Unified
Communications Manager identifies the need for a transcoder that is based on the capabilities
of the devices(G.711 only at the voice-mail system)and the maximum permittedcodec
(G.729). A device may support only a codec with higher bandwidth requirements than
permitted by the region configuration, for example. If such a device can access a transcoder. the
call is set up and invokes the transcoder resource. The call would otherwise fail.
3-16 Implementing Cisco Unified Communications Manager. Part 2 (CIPT2) v8.0 ) 2010 Cisco Systems, Inc
Example: Implementing a Transcoder at the Main Site
The figure illustrates an example of implementing a transcoder at the main site.
Cisco Unified
Communications
Manager
At the main site, there are two devicesthat supportCi.711 only. One device is a Cisco Unified
Communications Manager software conference bridge; theother device is a third-party voice-
mail application.
Regions are configured in such a way that all voice traffic between the remote site and the main
site has to use the G.729 codec.
When a user at a remote site needs to be added to a conference via the software conference
bridge, the user cannot be added, because G.729 must be used over the IP WAN but only G.711
is supported by the conference bridge.
By adding a transcoder resource at the main site gateway, you enable the remote site user to
send a G.729 voice stream, which is transcoded to G.711 and passed on to the conference
bridge b\ the transcoder that is located at the main site.
The same approach can beused for calls to the voice-mail system from the remote site.
3-14 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 ©2010 Cisco Systems, Inc.
Transcoder Implementation
This topic describes how to implement transcoders in order to allow low-bandwidth codecs to
be used when they are not supported by both endpoints.
As mentioned earlier in this lesson, transcoders arc devices that transcode voice streams, fhat
is. they changethe way that the audio payload is encoded(for instance, G.711 audio streams
are changed to G.729 audiostreams). Transcoders arc deployedin order to allow the use of
low-bandwidth codecs over the IP WAN even if one of the endpoints supports only high-
bandwidth codecs such as G.711.
The transcoder hasto be deployed close to the device thatsupports only G.711. Thatdevice
will send a G.711 stream to the transcoder. which transcodes the audio to a low-bandwidth
codec such as G.729. The G.729 voice stream is then sent from the transcoder to the other
device— a phone that is located at a remotesite—overthe IP WAN.
Note Itis important forthe MRGL to know that the devicethat is limited to the higher bandwidth
codec is the one that will request the transcoder media resource. If, for example, onlyG 729
is permitted between two IPphones, butone IP phone supports only G.711, the phone that
cannotcomply with the permitted codec (G.729, inthiscase) is the one that will requesta
transcoder. Therefore,the MRGL of this phone has to have access to a transcoder, which
should be physically located close to the requesting device. Regions have to be set up in
such a waythat the requesting phone is allowed to use G.711 to the transcoder(notethat
this call leg is also subject to region configuration).
Refore deploying transcoders. you must consider some factors that are like the factors that must
be considered when you deploy local conference bridges. Here arethe factors:
• Cost of adding DSPs: Is it necessary lo add DSPs to anexisting router only, or docs t le
whole platform have to be replaced?
) 2010 Cisco Systems. Inc Bandwidth Management and CAC Implementation 3-13
Example: Implementing Local
Conference Bridges at Two Sites (Cont.)
HQ_SW-
MRG
Main Site
Cisco Unified
Communications
Manager
The figure illustrates how Media Resource Groups (MRGs) and Media Resource Group Lists
(MRGLs) are used to ensure that headquarters phones use the conference resources at the
headquarters and that remote site phones use the remote site conference resource when
establishing a conference.
These three MRGs are created:
• HQ_HW-MRG: Includes the hardware conference bridge that is provided by the voice
gateway that is located at the headquarters
• IIQ_SW-MRG: Includes the software conference bridge that is provided by a Cisco
Unified Communications Manager server that is located at the headquarters
• BR_H\V-MRG: Includes the hardware conference bridge that is provided by the voice
gateway that is located at the remote site
The HQ HW-MRG is the first entry of the MRGL, which is called HQ_MRGL; (he HQ SW-
MRG is the next entry. Headquartersphones are configured with the HQ_MRGL. Because
MRGs arc used in a prioritized way, headquarters phones that invoke a conference will first use
the available hardware conference resources; when all of them are in use, the software
conference resources are accessed.
At the remote site, all phones refer to the BRMRGL, which includes only the BRJIW-MRG.
This configuration allows remotephonesto use their local conference bridge when they invoke
conferences instead of accessing conference resources that are located across the IP WAN.
3-12 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 J2010 Cisco Systems, Inc.
Example: Implementing Local Conference Bridges at Two Sites
The figure shows a sample scenario for deploying local conference bridge resources at a remote
site.
Main Site
Cisco Unified
Communications
Manager
WAN
Remote Site
The figure shows a main site with software and hardware conference resources. At the remote
site, hardware conference resources are added lo the remote site gateway. As a result, the
remotesite phones can set up conferences by using local resources insteadof by always
accessing the conference resources that are located at the main site. For conferencing remote
site members onlv. no traffic has to be sent across the IP WAN.
Note When an ad hoc conference includes members of separate sites, a separate voice stream
for each remote member has to be sent across the IP WAN. However, if a Meet-Me
conference is set up, the users that are located at the remote site could first establish an ad
hoc conference (by using a media resource that is localto the remote users) and then add a
call to the remote Meet-Me conference to their local ad hoc conference. In this case, there is
onlya single voice stream that is sent across the IP WAN connecting the twoconferences.
When local conference bridges or Media Termination Points (MTPs) are deployed at each site,
traffic does not have to cross the IP WAN if all endpoints are located at the same site. You can
implement local media resources such as conference bridges and MTPs by providing
appropriate hardware (digital signal processors [DSPs]) at the routers that are located at the
remote sites.
Whether the extra cost for providing the DSP resources will be worthwhile depends on several
factors:
• Cost of adding DSPs: Is it necessary to add DSPs to an existing router only, or does the
whole platform have to be replaced?
• Number of devices at remote site and likelihood of using applications or features that
require access to the media resource that is considered to be locally deployed: How
many phones are located at the remote site? How often do the phones use features that
require a media resource that is currently available only over the IP WAN? What is the
maximum number ofdevices that require access to the media resource at the same time?
• Available bandwidth and cost of additional bandwidth: Is (here enough bandwidth (or
can additional bandwidth be provisioned) to accommodate the requirements that are
determined by the preceding factors? How does the cost of adding bandwidth compare to
the cost of deploying local DSPs?
3-10 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 J2010 Cisco Systems, Inc.
Example: Codec Configu ration (Cont)
ftevw JnfonmHIon————
flimt E4L_0|.2ftai
•teuton RalBilQn4jiw* - - -
*mli*0 Hm,*mm+****t*
5=_J- - .bpif^ -si.
"fhe figure illustrates region configuration in Cisco Unified Communications Manager for the
discussed scenario. The configuration of the HQ_phones and the BR_phones regions is
illustrated. Both regions are configured in such a way that calls within the region and calls to
the local gateway (regions HQ_gw and RR_gw) arc allowed to use (i.7l I. while calls to all
other regions are limited to G.729.
Note The preceding example is a partial configuration only. It does not show the configuration of
the other regions.
Region Configuration
WitntnHQ_gw G711
HQ_ptionestoHQ_gw G.711
Remote Site
In the figure, phones that are located in the headquarters are configured with region
HQ_phones. An intercluster trunk that connects to another Cisco Unified Communications
Manager cluster and a Session Initiation Protocol (SIP) trunk connecting to an Internet
telephony service provider (ITSP) are in region HQ_trunks. The public switched telephone
network (PSTN) gateway that is located in the headquarters is configured with region HQ_gw.
At the remote site, phones are in region BR_phones and the PSTN gateway is in region
BR_gw.
Cisco Unified Communications Manager regions are configured in the following way:
Within HQ_phones: G.711
Within IIQ_gw: G.711
HQ_phones to HQ_£w: G.711
Within BR_phones: G.711
Within BR_gw: G.711
BR_phones to BR_gw: G.711
All others: G.729
As a result, of this configuration, all calls that use the IP WAN between the remote site and the
headquarters use G.729. Calls that are sent through the intercluster or SIP trunk use G.729 as
well. These calls use G.711: calls between phones within the headquarters, calls between
phones within the remote site, calls from headquarters phones to the headquarters PS'fN
gateway, and calls from remote site phones to the remote site PSTN gateway.
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 >2010 Cisco Systems, Inc.
Review of Cisco Unified Communications Manager Codecs
This subtopic reviews how to control the codec that is used for a call in Cisco Unified
Communications Manager.
The codec that will be used for a call depends on the Cisco Unified Communications Manager
region configuration. Eachdevice is assigned with a regionvia the device pool configuration.
For each region, the administrator can configure the highestpermitted codec bandwidth within
a region, to other specifically listed regions, and to all other (not listed) regions.
Whena call is placed between two devices, the codec is determined based on the regionsof the
two devices and on the capabilities of the devices: The devices will use the bestcodec that is
supported by both devices and that doesnotexceed the configured codec bandwidth forthe
region or regions that are involved in the call. If the two devicescannotagree on a codec (for
instance, if region configuration allowsonly 8 kb/s as the maximum codec bandwidth but one
device supports only G.711). a transcoder is invoked, if available. The losstypeof a linkcan
alsobe configured. On links that are configured to be lossy, codecs that are lesssensitive to
packet lossarepreferred overcodecs thatresult in higher-quality degradation. Formore details
about codec selection, refer to the Implementing Cisco Unified Communications Manager,
Part / (CIPTI) course.
To conserve IP WAN bandwidth, you should use low-bandwidth codecs in the IP WAN. For
calls within a LAN environment, you should use high-bandwidth codecs for optimal audio
quality. Whenyou are designing where to use which type of codec, it is important to consider
that low-bandwidth codecs such as G.729are designed for human speech. They do not work
well for other audio streams, such as music.
As stated in the previous topic, other methods exist for limiting the bandwidth that is required
for MOH streams. If you cannot use multicast MOH from branch router flash but MOH streams
are not desiredon the IP WAN. you can disable MOHfor remotesite phones.
3-6 Implemenling CiscoUnified Communications Manager, Part 2 (CIPT2) v8,0 ) 2010 Cisco Systems, Inc.
Other bandwidth management solutions includethe use of transcoders or the implementation of
special features such as multicast MOH from branch router flash. Transcoders are devicesthat
can transcode voice streams. That is. they change the way that the audio payload is encoded
(for instance, a G.711 audio stream is changed to a G.729 audio stream). Transcoders allow the
use of low-bandwidth codecs over the IP WAN even if one of the endpoinls is limited to a
high-bandwidth codec such as G.711. Multicast MOII from branch router Hash allows a
multicast MOH stream lo be generated by a Cisco IOS router that is located at the remote site,
instead of being sent over the IP WAN from a centralized MOH server.
- Transcoders
Note Refer to the "Quality of Service" moduleof the Implementing Cisco Voice Communications
and QoS (CVOICE) course for more detailed discussion about QoS.
Otheroptions for managing IP WANbandwidth are techniques that influence where voice
streams are sent. If three phones, all located at a remotesite, establishan ad hoc conference,
there is a greatdifference in bandwidth usage if the conference bridge is located at that remote
site—local to the phones that are members of the conference—or if the conference is located at
the main site and has to be accessed over the IP WAN. Inthe latter case, all three phones are
sending their voice stream tothe conference bridge over the IP WAN. The conference bridge is
mixing thereceived audio and is then streaming it back to all conference members (in three
separate streams). Although the call appears to be local to the remote site—because all
conference members are located atthat site—due tothe remotely located conference bridge, the
IP WAN is occupied by three calls.
3-4 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 © 2010 Cisco Systems, Inc.
Lesson 1
Managing Bandwidth
Overview
When an IP WAN connects various sites in a Cisco Unified Communications network,
bandwidth consumption at the IP WAN should beminimized. Several techniques can help
conserve bandwidth on the IP WAN in a multisite deployment;
• Reducing the required bandwidth of voice streams
• Keeping some \oice streams (such as local media resources) away from the IP WAN
• Employing special features like multicast music on hold (MOH) from branch router flash
(or the use of transcoders).
"fhis lessondescribes all these techniques and features and their implementation.
Objectives
Upon completing this lesson, you will be able to describe techniques to reduce bandwidth
requirements on IP WAN links in Cisco Unified Communications Manager multisite
deployments. This ability includes being able tomeet these objectives:
• Describe methods to minimize bandwidth requirements for Cisco Unilied Communications
• Configure Cisco Unified Communications Manager in order tocontrol the codec that is
used for a call
• Implement local conference bridges in order to avoid accessing conference bridges over the
IP WAN even if all participants are local
• Implemeni transcoders in order to allow low-bandwidth codecs to be used iflow-bandwidth
codecsare not supported by both endpoints
• Implement multicast MOH from branch router flash to avoid MOH streams over the IP
WAN
3-2 Implementing Cisco Unrfied Communications Manager, Part 2(CIPT2) v8.0 ©2010 Cisco Systems, inc.
Module 3
Module Objectives
Upon completing this module, you will be able to implement bandwidth management and CAC
to prevent oversubscription ofthe IP WAN. This ability includes being able to meet these
objectives:
• Describe techniques to reduce bandwidth requirements on IP WAN links in Cisco Unified
Communications Manager multisite deployments
• Describe and configure CAC mechanisms and AAR in Cisco Unified Communications
Manager and in gatekeepers
Module Self-Check Answer Key
QD C
02) B. C
05) D
06) B
Q7) B
Q8) D
09) D
010) A
Qll) D
Q12) B.C
Q13) A,D
Q14) B.C
Q15) A
2-102 Impiementing Cisco Unified Communications Manager, Part 2 (CIPT2) vB.O ©2010 Cisco Systems, Inc.
Q12) Which two statements about Cisco Unified Communications Manager Express are
true? (Choose two.) (Source: Implementing Cisco Unified Communications Manager
Express in SRST Mode)
A) IP phonesregisterwith Cisco Unified Communications Manager Express in
standalone mode when Cisco Unified Communications Manager Express is
partof the Cisco Unified Communications Manager group (hat is specified in
the device pool of the phone.
B) During SRST fallback. IP phonesregisterwith Cisco Unified Communications
ManagerExpress in SRSTmode when Cisco Unified Communications
ManagerExpress is configured as the SRST reference for the IP phone.
C) Cisco Unified Communications Manager Express in SRST modeprovides
more features than standard SRST.
D) The same platform can serve more phones when running Cisco Unified
Communications Manager Express in SRST mode versus running standard
SRST,
E) Standalone Cisco Unified Communications Manager Express routerscan be
clustered for redundancy.
013) Which two features have been added in Cisco Unified Communications Manager
Express Release 8.0? (Choose two.) (Source: implementing Cisco Unitied
Communications Manager Express in SRST Mode)
A) fi\ e additional MOI 1sources
B) presence with BLE status
C) \ideo support
D) demote argument of the dialplan pattern command
E) local MOII
Q7) The SRST reference is configured under System > Enterprise Phone Parameters.
(Source: Implementing SRST and MGCP Fallback)
A) true
B) false
Q8) Which command is used for SRST configuration at the Cisco IOS router? (Source:
Implementing SRST and MGCP Fallback)
A) telephony-server
B) ccm-manager fallback
C) service alternate default
D) call-manager-fallback
Q9) Which command is not used for MGCP Fallback configuration? (Source:
Implementing SRST and MGCP Fallback)
A) ccm-manager fallback-mgcp
B) application
C) global
D) telephony-server
E) service alternate default
QIO) Wheredo you configure the maximumnumberofhops that can be used by CFUR?
(Source: Implementing SRST and MGCP Fallback)
A) service parameter
B) enterprise parameter
C) SRST gateway configuration
D) phone configuration
Q1I) How cancalling privileges be implemented for SRST individual phones? (Source:
Implementing SRST and MGCP Fallback)
A) only when using Cisco Unified Communications ManagerExpress in SRST
mode
B) by preconfiguring the phonesthat need callingprivileges assigned
C) by configuring an ephone-dn template
D) by configuring COR lists for directory numbers
2-100 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 ©2010 Cisco Systems, Inc.
Module Self-Check
Use the questions hereto review whatyou learned in this module. Thecorrect answers and
solutions are found in the Module Self-Check Answer Key.
Q2) Which two types of calls arc notpreserved during switchover of an SRSf gateway?
(Choose two.) (Source: Examining Remote Site Redundancy Options)
A) calls between IP phones that are located at the remote site
B) conference calls of remote-site phones using a conference bridge that is located
at the main site
C) calls between IP phones that arc located at the remotesite and at the main site
D) calls between IP phones that are located at the main site
E) calls from main-site phonesthat wereplaced to a remote-site phone and then
transferred from the remote-site phone to another main-site phone
Q3) Which configuration is required lo allow calls lo bepreserved during switchback from
H.323 to MGCP? (Source: Examining Remote Site Redundancy Options)
Q4) What are the two correct statements of supported phones in SRST for the given
platform? (Choose two.) (Source; Examining Remote Site Redundancy Options)
A) 800: 4
B) 2801:500
C) 2851:350
D) 3825:350
E) 3845:1024
Q5) What can you use toconfigure the dial plan ata remote-site gateway insuch a way that
branch users can stillreach the headquarters when dialing internal directory numbers
during fallback? (Source: Examining Remote Site Redundancy Options)
A) Ihis is not possible. Users have to dial headquarters users by their PSTN
numbers while in fallback mode.
B) Use translation profiles modifying the callingnumber.
C) Issue the dialplan-pattern command.
D) Use translation profiles modifying the called number.
2-9S Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 ©2010 Cisco Systems, Inc.
Module Summary
This topic summarizes the key points that were discussed in this module.
Module Summary
Ihis module described the available features for providing remote phones with backup in the
event of an IP WAN outage. It explained call survivability. Media Gateway Control Protocol
(MGCP) fallback, and Cisco Unified Survivable Remote Site Telephony (SRS'f). fhe module
also contrasted the differences between standard SRST and Cisco Unified Communications
Manager Express in SRST mode. In addition, itdescribed how to implement standard SRST
and a dial plan to support intersite connectivity through the public switched telephone network
(PSTN), as well as PSTN access during IP WAN failure. Finally, the module showed how to
implement a backup solution using Cisco Unified Communications Manager Express in SRST
mode instead of using standard Cisco Unified SRST.
References
For additional information, refer to these resources:
• Cisco Systems. Inc. Cisco Unified Communications System 8.x SRND, April 20IC.
hup://\\\\w,cisco.com •'en.TJS/d()cs/voicejp_comni/ciiein/snid/8x/uc8x.hlinl
• Cisco Systems. Inc. Cisco Unified Communications Manager Administration Guide
Release 8.011). February 2010.
http://www.cisco.coni/cn/t'S/docs/v()icejp_coinm/cLicni/admin/8J) l/ecmcfg/bccm-KOI-
cm.html
• Cisco Systems. Inc. Cisco Unified Survivable Remote Site Telephony Version 8.0.
November 2009,
lntp://w\\w.cisco.eoni/en/US/prod/collateral/voicesxv/ps6788/vcallcon/ps2l69/daUi .sheet_>
78-570481.html
Summary
References
For additional information, refer to this resource:
• Cisco Systems. Inc. Cisco Unified Communications Manager Express System
Administrator Guide. November 2007 with updates 2010.
http:'•'www.cisco.coiTt/en.-rS/docs.'Voice ip coiiim-'ciiciiie/adniin/cunliguration/guide/cnica
dm.html
Cisco
Unity
Cisco Unified
Communications
Manager
telephony-service
erst mode Auto-provision none
srst dn line-mode dual
srst aphone templata 1
srst dn template 3
srst ephone description CUCME-SRST
ephone-template 1
keep-conference local-o Qly
ephone-dn-template 3
hold-alert 25 idle
In the example. Cisco Unified Communications Manager Express uses ephone template 1 for
newly added phones. This template configures conferences to drop if no internal members are
left in the conference.
Ephone-dns, which are learned using SNAP, are configured to alert the user if a call is on hold
for 25 seconds and the phone is idle.
For easier distinction, the description of learned phones should include an SRS'f string. The
ephone-dns should be dual-mode lines.
2-94 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) w8.0 ) 2010 Cisco Systems, Inc.
Note Ifsingle-line ephone-dns is used with multiline features likecall waiting, Call Transfer, and
conferencing, there must be more than one single-line directory number on a phone.
CHERouter(confia-telephonv)#
srst dn template template-tag
CHERouter(config-telephony)#
srst ephone template template-tag
CHERouter(config-telephony)#
srst mode auto-provision {all | dn | none}
• Enables SRST mode for a Cisco Unified Communications Manager
Express router
- all writes information for leamed ephones and ephone-dns into
the running configuration.
- dn writes information for learned ephone-dn into the running
configuration.
- none does not include information for learned ephones or
ephone-dns in the running configuration.
CHERouter(confiq-telephony)#
srst dn line-mode {dual | single}
• Specifies the line mode for ephone-dns in SRST mode on a Cisco
Unified Communications Manager Express router. Default is single
line
To enable SRS'f mode for Cisco Unified Communications Manager Express, use the srst mode
auto-provision command in telephony-service configuration mode:
• The keywordall includes information for leamed ephonesand ephone-dns in the running
configuration.
Note Ifthe administrator saves the running configuration after learning ephones and ephone-dns,
the fallback IP phones will be treated as locally configured IP phones on the Cisco Unified
N
Communications Manager Express SRST router,whichcould adversely impact the fallback
behavior of those IP phones.
To specify the line mode for the ephone-dns that are automatically created in SRST mode on a
Cisco Unified Communications ManagerExpress router,use the srst dn line-mode command
in telephony-service configuration mode. The keywords provide these specifications:
• The keyword dual specifies dual-line ephone-dns.
• The keywordsingle specifies single-line ephone-dns (the default).
2-92 Implementing Cisco Unified Communications Manager,Part 2 (CIPT2) vB.O >2010 Cisco Systems, Inc.
Phone Registration Process
This section describes the phone registration process.
When a phone loses connectivity to the Cisco Unified Communications Manager, itregisters to
its configured SRST reference.
If that SRST reference is Cisco Unified Communications Manager Express in SRS'f mode, the
Cisco Unified Communications Manager Express router firstsearches for anexisting
(preconfigured) ephone with the MAC address ofthe registering phone. Ifthe router finds an
ephone. the stored ephone configuration isused. No phone configuration settings that arc
pro\ ided b> SNAP are applied, and no ephone template is applied. Ifthe configured ephone is
configured with one ormore ephone-dns. the stored configuration isused for the ephone-dn or
ephone-dns ofthe phone. Neither the information that is provided by SNAP nor the ephone
template thai isconfigured under telephony-service isapplied. Ifthe configured ephone isnot
configured with an ephone-dn. automatic assignment has tobe enabled for the phone tobecome
associated with an ephone-dn. SNAPis no option in this case.
Ifno ephone is found for the MAC address ofthe registering phone. Cisco Unified
Communications Manager Express adds the ephone (and applies the ephone template, if
configured), using SNAP. Ifthe directory number exists, it is bound tothe added phone:
otherwise, the directory number islearned using SNAP. Ifconfigured, the ephone-dn template
is applied.
These advantages allow flexible configuration of any Cisco Unified Communications Manager
Express features in a scalable way, since only those phones and directory numbers that require
additional features (or individual settings) have to be manually preconfigured.
2-90 ImplementingCisco Unified Communications Manager, Part 2 (CIPT2)v8.0 ) 2010 Cisco Systems, Inc.
Phone Provisioning Options
The table summarizes the phone provisioning options and shows the relevant configuration
parts.
As shown in the table, if an ephone and ephone-dn are configured in Cisco Unified
Communications Manager Express, a phone that registers with the configured MAC address
will get the complete configuration (phone and director;' number) appliedas configured in
Cisco Unified Communications Manager Express. Cisco Unilied Communications Manager
Express does not use SNAP at all to configure the phone.
If an ephone is configured butis notassociated with an ephone-dn, automatic assignment hasto
be enabled. Otherwise, the phone will not have a line and cannotplace or receive calls. The
ephone-dn configuration is determined based on the arguments of theauto-assign command.
SNAP is not used (for learning phone settings or director,' number configuration parameters).
If only ephone-dns are configured, the ephone configuration is learned by SNAP, while the
ephone-dn configuration that is configured in Cisco Unified Communications Manager Express
is used instead of the phone directory number configuration that is provided by SNAP. Ephone
templates (if configured) areapplied to the learned ephone configuration.
If neitheran ephone (MAC address) nor a directory numberexists for the registering phone.
Cisco Unified Communications Manager Express will learneverything(ephone and ephonc-dn
configuration) from SNAP. Ephone and ephone-dn templates are applied, if configured.
2-88 Implementing Cisco Unrfied Communications Manager, Part 2 (CIPT2) v8.0 © 2010 Cisco Systems. Inc
Configuration of Cisco Unified Communications
Manager Express in SRST Mode
This topic describes how to configure Cisco Unified Communications Manager Express in
SRST mode.
voice moh-group 1
moh Elashimohl.au
description HOH: customer services
multicast moh 239.1.1.1 port 16381
extension-range 1000 Co 1099
extension-range 1300 to 1399
I
voice moh-group 2
moh flashimoh2.au
description HOB: marketing •r
multicast moh 239.1.1.2 port 16384
extension-range 3000 to 3099
I
telephony - service
moh-file-buffer 5000
moh flashidefault.wav
multicast moh 239.1.1.3 port 163 64
For each department, an MOH group is configured. Within each group, the location of the
MOH audio file and the extensions that should utilize the group have to be configured. In
addition, an optional description can be configured and multicast MOH can be enabled for each
MOH group.
You configure RAM caching under call-manager-fallback (in the case of Cisco Unified SRST)
or under telephony-service (in the case of Cisco Unified Communications Manager Express).
You use the moh-file-buffer size-in-kb command for this configuration, and it specifies the
maximum size of the MOH RAM cache, 'fhe configured limit applies to each audio source file.
You cannot enable or disable audio source caching on a per-file basis. The total amount that is
used for audio source caching, therefore, dependson the numberof configured MOI 1groups. If
all five possible MOH groups and a default audio source are configured, the file buffer size that
is allocated will be six times the specifiedamount. Ifa configured audiosource file is larger
than the configured moh-file-buffer, it will not be cached but will be read from flash instead.
Note You can use the show flash command to see the size of the MOH files.
2-86 Implementing Cisco Unified Communications Manager. Part 2 (CIPT2) v8.0 >201u Cisco Systems, Inc.
*m»
Additional MOH Sources
Cisco Unified Communications Manager Express Version 8 introduces the capability of
•Mr configuring up to five MOH sources in addition to the default MOH source.
These additional MOH sources can be utilized by SCCP phones that put calls on hold. Any
other entities thai putcallson hold (such as basicautomatic calldistribution [B-ACD] or SIP
phones) will use the default MOH source. If live audio feed is used, it can beconfigured only
as the default MOH source.
Cisco Unified Communications Manager Express can be configured to cache files in RAM.
This configuration reduces CPU utilization because flash reads areessentially eliminated after
the audio fileshave been loadedto RAM. However, cachingaudio files in RAM can drastically
increase memory consumption. Memory requirements depend onthenumber of MOI I files and
their size (there is no limitation on the maximum size of an audio file).
Multiple MOH sources arc supported by Cisco Unified Communications Manager Express.
Cisco Unified Communications Manager Express in SRSTmode, and Cisco Unified SRST.
Multiple MOH sources arc supported onthese platforms: Cisco Unified Communications 500
Series and Cisco 1800. 2800. 2900, 3800. and 3900 Series Integrated Services Routers.
The audio files have to be .au or .wav files in G.711 8-bit mono format, and their minimum size
is 100 kb. Ifmultiple flash devices are present inthe router, the default flash drive should be
utilized.
The configuration of multiple MOH sources isbased on MOH groups. Endpoints that do not
support MOH groups orthat are not configured to use an MOH group will use the default MOH
source.
Note The MOH source is selected based on the configuration of the holder (thatis, the phonethat
puts the call on hold).
^^hr
-km*
When thephone thatis receiving MOH is part of a system thatuses a G.729 codec, transcoding
is required between G.711 and G.729. TheG.711 MOH must betranslated to G.729. Note that,
because of compression. MOH that is using G.729 is of significantly lower fidelity than MOII
that is using G.711.
If the MOH audio stream is also identified as a multicast source, the Cisco Unified
Communications ManagerExpress router additionally transmits the streamon the physical IP
interfaces of the Cisco Unified Communications Manager Express router that you specify
during configuration. This transmission permitsexternal devices lo have accessto the router.
Certain IP phones do notsupport IP multicast and, therefore, do notsupport multicast MOI I.
You can disable multicast MOII to individual phones that do not support multicast. Callers hear
a repeating tone when they are placed on hold.
In Cisco Unified Communicalions Manager Express, the MOH feature is supported when a call
is put on hold from a SIP phone and when the user of a SIP phone is put on hold by a SIP,
SCCP. or plain old telephone service (POTS) endpoint. The holder (the party who pressed the
Hold key) or holdee (the party who is put on hold) can be on the same Cisco Unified
Communications Manager Express group or on a different Cisco Unified Communications
Manager Express group that is connected through a SIP trunk. MOH is also supported for Call
Transfer and conferencing, with or without a transcoding device.
Configuring MOH for SIP phones is the same as configuring MOH for SCCP phones.
2-84 Implemenling Cisco Unified Communications Manager, Pan 2 (CIPT2) v8.0 ) 2010 Cisco Systems, Inc
Providing Phone Loads
You can configure Cisco Unified Communications Manager Express to provide specific phone
loads to IP phones for each type of phone.
Make them
accessible by TFTP.
Specify the phone
load to be used per
tftp-aerver flash; apps4S.9-Q-2ES2.sbn
phone type under tftp-server flash; cnu 4 5.9 - 0 - 2 ES2.sbn
telephony service. tftp-server flash; cmv45sccp.9-0-2ES2.sbn
tftp-server flash: dsp45.9-0-2ES2.sbn
If no phone load is
tftp-server flash: jar45sccp.9-0-2BS2.sbn
specified, the current tftp-server flash: SCCP4 5.9-0-2SR1S.loads
phone load of the tftp-server flash: term45.default.loads
phone is used. tftp-aerver flash: termS5.default.loads
I
telephony-service
load 7965 SCCP45. 9-0-2SR1S
load 7945 SCCP45. 9-0-2SR1S
Note You can view a list of IP phone models that are supported by the Cisco Unified
Communications Manager Express router by entering the load ? command in telephony-
service configuration mode
ephone 3
mac-address 0012.0154.5D98
type 7 96 0
button 1:6
ephone 4
mac-address 0007.0E5 7.6F43
type 7961
button 1:7
1
dlalplan-pattern 3 5215553... extension-length 4
2-82 Implementing CiscoUnrfied Communicalions Manager, Part 2 (CIPT2) v8.0 J2010 Cisco Systems, Inc.
number: Use this command lo define a directory number for an ephone-dn (extension),
which then can be assigned to an IP phone.
ephone: Use thiscommand in global configuration mode to create a phone in Cisco
Unified Communications Manager Express.
mac-address: Use this command in ephone configuration mode to specify the device ID of
an ephone. When a phone registers with Cisco Unified Communications ManagerExpress,
it has to providea device ID (which is based on the MAC addressof the phone) that is
configured in Cisco Unified Communications ManagerExpress.
type: Use this command in ephone configuration mode lo specify the phone type of this
ephone.
button: Use this command in ephone configuration mode to assign one or moreephor>
dnsto an ephone.
dialpian-pattem: Use thiscommand in telephony-serv ice configuration mode to map
E. 164 PSTN numbers to internal extension numbers.
Note The default values of max-ephones and max-dn are 0. These defaults have to be modified
inorder foryouto configure ephones and ephone-dns. The maximum numberof supported
ephones and ephone-dns is version-specific and platform-specific. The number that is
displayed in Cisco IPS Software Help files does not always reflect the actual limit.
2-80 Implementing Cisco Unrfied Communications Manager, Part 2 (CIPT2) v8.0 >2010 Cisco Systems, Inc.
Important Cisco Unified Communications Manager Express
Features
Cisco Unilied Communications Manager Express also includes features that are like legacy
low-end PBXand key system features, creatinga cost-effective, highly reliable, feature-rich
communications solution for the small office.
Ihe figure lists important features of Cisco Unified Communications Manager Express.
fhese new features are introduced with Cisco Unified Communications Manager Express
Version 8.0:
• Kive additional music on hold (MOH) sources: SkinnyClient Control Protocol (SCCP)
phones can beconfigured to use oneof five additional MOII source files thatarc
configured by MOH groups.
• Support for fc.164 numbers and + prefixes: IP phones can use E.164 format with a +
prefix for their directory numbers.
• Enhancement of the dialplan pattern command: You can use the dialplan pattern
command to allow internal devices to call each other by an internally used shorter number
that is derived from a longer directory number of the phone {typically in E.164 format with
+ prefix).
• Enhancement of voice translation profiles: Whena cal! is sent to an IP phone,an
additional number (that is. a callback number) is sent to thephone. The phone shows the
calling-party number on its display but uses the callback number for callbacks from call
lists.
2-78 Implementing Cisco Unified Communications Manager, Part2 (CIPT2) v8.0 J2010 Cisco Systems, Inc.
In summary, use Cisco Unified Communications Managerwhen Cisco Unified
Communications Manager Express does not scale to the number of endpoints or does not
provide all therequired features. If youuse Cisco Unified Communications Manager and the
standard Cisco Unified SRST features do not meet the requirements for backup scenarios, you
should use Cisco Unified Communications Manager Express in SRST mode.
2-76 Implementing CiscoUnified Communications Manager, Part 2 (CIPT2) v8.0 >2010 Cisco Systems, Inc.
Because of the centralized architecture of Cisco Unified Communications Manager, remote site
survivability isextremely important. As discussed earlier. Cisco Unified SRST can be used to
provide survivability. However, it is quite limited in terms of telephony features.
To provide a richer feature that issetto IPphones that are in fallback mode, you can use Cisco
Unified Communications Manager Express in SRST mode. Such a deployment combines the
advantages ofCisco Unified Communications Manager—centralized configuration and the
availability offeatures to all phones, with the better feature support dial isprovided by Cisco
Unified Communications Manager Express versus standard Cisco Unified SRST in casethe site
is disconnected from the centralized Cisco Unilied Communications Manager cluster.
Clustering Yes No No
Centralized call
processfig
Yes No Only within local site
2-74 Implementing CiscoUnified Communications Manager, Part 2 (CIPT2) v8.0 © 2010 Cisco Systems, Inc.
Cisco Unified Communications Manager Express in SRST
Mode
SRSTfallback supportusing Cisco Unified Communications ManagerExpress is a leaturethat
enablesrouters to providecall-processing support for Cisco Unified IP phones if they lose
connection to remote primary, secondary-, or tertiary Cisco Unified Communications Manager
installations or if the WAN connection is down.
Cisco
Unity
H 323 or SIP
Trunk Connection
Cisco Unified
Communications
Regislrafoon Manager
and Signaling,' J
The figure shows a deployment of a Cisco Unified Communications Manager Express router
with several phones and devices that are connected to it. The Cisco Unified Communications
Manager Express router is connected to thepublic switched telephone network (PSTN) and
WAN.
2-72 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 >2010Cisco Systems, Inc
Lesson 3
Objectives
Upon completing this lesson, you will be able to configure Cisco Unified Communications
Manager Express to provide telephony services lo IP phones ifthe connection to the centralized
call agent islost. This ability includes being able tomeet these objectives:
• Describe Cisco Unified Communications Manager Express and the modes in which it can
be used
• Describe Cisco Unified Communications Manager Express versions, their protocol support,
their features, and the required Cisco IOS Software releases
• Describe general Cisco Unified Communications Manager Express configuration
parameters and their functions
• Describe how to configure Cisco Unified Communications Manager Express to support
SRSI fallback
Summary
This topic summarizes the key points that were discussed in this lesson.
Summary
References
For additional information, refer to these resources:
• Cisco Systems. Inc. Number Translation Using Voice Translation Profiles, February 2006.
http://\vw u.cisco.com,'en/US/teclL/tk652/lk90/teehnoloaies configuration e\ample()9186a0
0803f818a.shtmi
2-70 Implementing Cisco Unified Communications Manager, Part2 (CIPT2) v8.0 ) 2010 Cisco Systems, Inc.
Cisco Unified SRST Dial Plan Example
(Cont)
destination-pattern 91
port 1/0:23
Outgoing COR lists are applied to the outbound dial peers. Note that all dial peers that should
be available to all phones (that is. dial peer 911 for emergency and dial peer 2000 for intersite
calls) are nol configured with an outgoingCOR list.
Local and national PSI'N destinations areprotected by outgoing COR listlocal-ntl. This COR
has one member, pstn-local-ntl. and this member islisted only in the incoming COR list of
Phonc2. not in the incoming COR listof Phone3. Dial peer 9011, which is used for
international calls, is configured with an outgoing COR list intl, and the only member ofthai
list, pstn-intl. is not included in the incoming COR lists ofPhone2 and Phone3.
One dial peer is configured with ihe incoming ealled-numbcr . command, "fhis dial peer is
used as an incoming POTS dial peer, fhe dial peer isconfigured tosupport direct inward
dialing.
The called numbers of inbound PSTN calls (521 555-3xxx) are mapped to four-digit extensions
because ofthe dialplan-pattern command that is configured in call-manager-fallback
configuration mode (see earlier in this subtopic). As aresult, incoming PSTN calls are sent to
the four-digit extensions.
Outgoing calls to phones that are located at the main site (calls to 2...) match the destination
pattern in dial peer 2000. That dial peer sends calls to port 1/0:23 after performing digit
manipulation using the to-HQ voice translation profile, 'fhis profile translates the four-digit
calling number to an Il-digit E.164 PSTN number, which means that during SRST fallback,
users can still dial 4-digit extensions to reach the headquarters.
km*
Thefigure shows the first part of the SRST configuration. Itincludes a dialplan-pattern
command (configured in call-manager-fallback configuration mode) thatmaps the internal
four-digit directory numbers to the E.164 PSTN number.
Based on the scenario, one phone (Phonel) should have unlimited access. No incoming COR
list is required at that phone because, in the absence of an incoming CORlist, all outbound dial
peers are available regardless of a configured outgoing COR listat theoutbound dial peer.
The other two phpnes should have difTerent classes; therefore, anincoming COR listis
configured for each of them (COR lists Phone2 and Phone3). Phone3 should not be allowed to
dial the PSTN at all (exceptfor emergency calls to 911), while Phone2 should not be allowedto
dial international PSTN destinations.
The dial peer that will be used for emergency calls will not be configured with an outgoing
COR list, and hence will beavailable to all callers. Thesame principle applies to all internal
directory numbers. Because they are not configured with anoutgoing COR list, they all are
reachable by everyone. The dial peer for international calls will be protected with outgoing
COR list intl. The member ofthis outgoing COR list (pstn-intl) isnot listed in the incoming
COR list ofeither Phone2 or Phone3. This way, neither ofthese phones can place international
calls. Asmentioned earlier. Phonel does nothave an incoming COR list, and therefore, the
outgoing COR list at the international dial peer isignored for calls from Phone 1, Finally, all
other PSTN dial peers (local and national calls) are protected with outgoing COR list local-ntl.
The incoming COR list ofPhone2 includes the member ofoutgoing COR list local-ntl (pstn-
local-ntl) and therefore can dial local and national PSTN destinations but is not able to dial
internationally. The incoming COR list ofPhone3 includes a member that iscalled no-pstn that
isnot listed in any outgoing COR list. Ilence, this incoming COR list does not provide access
to any protected pattern. Its only use is to change from the default behaviorthat in the absence
ofan incoming COR list all outgoing COR lists are ignored and hence all outbound dial peers
are available. You could also configure COR list Phone3 with no member, and itwould have
the same effect. Ilow ever, it isrecommended that you always include at least one member ner
COR list.
2-68 Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 >2010 Cisco Systems, Inc.
Cisco Unified SRST Dial Plan Example
The figure shows an example ofa standalone dial plan configuration for a Cisco IOS router that
is enabled with Cisco Unified SRST.
Cisco
Unified
Phonel
Communications
Manager
Phone2
Phone3
The example shows a headquarters site with a PSI'N number of511 555-2xxx, and a remote
site witha PSTN number of 521 555-3xxx. four digits are used for internal calls(including
calls between the main site and remote site).
There are three phones atthe remote site. During SRS'f fallback. Phonel (using directory
number 3001) should have unlimited access. Phone2 (directory number 3002) should not be
allowed toplace international calls, and Phonc3 (directory number 3003) should be allowed to
place only internal calls. Four-digit dialing to the headquarters should work: the calls should be
sent to the main site over the PSTN.
No COR COR list that is Call will succeed. By default, the incoming dial peer
applied for outgoing has the highest COR priority when no COR is
calls applied. Ifyou apply no COR for an incoming call
leg to a dial peer, the dial peer can make a call out
of any other dial peer, regardless of the COR
configuration on the outgoing dial peer.
COR list that is applied No COR Call will succeed By default, the outgoing dial peer
for incoming calls has the lowest priority. Because there are some
COR configurations for incoming calls on the
incoming or originating dial peer, it is a superset of
the outgoing-call COR configuration for the
outgoing or terminating dial peer.
COR list that is applied COR list that Is Call will succeed. The COR list for incoming calls
for incoming calls applied for outgoing on the incoming dial peer is a superset of the COR
(superset of COR list calls (subsets of COR list for outgoing calls on the outgoing dial peer
that is applied for list that is applied for
outgoing calls on the incoming calls on the
outgoing dial peer) incoming dial peer)
COR list that is applied COR list that is Call will not succeed. The COR list for incoming
for incoming calls applied for outgoing calls on the incoming dial peer is not a superset of
(subset of COR list calls (supersets of the COR list for outgoing calls on the outgoing dial
that is applied for COR list that is peer.
outgoing calls on the applied for incoming
outgoing dial peer) calls on the incoming
dial peer)
2-66 Implementing Cisco Unified Communications Manager, Part2 (CIPT2) v8.0 )2010 Cisco Systems, Inc.
Cisco Unified SRST Dial Plan Commands: COR
You can assign calling privileges to IP phones when they are in SRS'f mode by using COR
commands.
router(config-cm-fallback)#
cor {incoming I outgoing} cor-list-name [cor-list-number
starting-number - ending-number | default]
• Configures a COR on dial peers associated with directory
numbers (ephone-dn)
The command cor configures a COR ondial peers that are associated with directory numbers.
• The keyword, incoming specifies that the COR list is tobe used by incoming dial peers.
• The keyword outgoing specifies that the COR list is tobe used by outgoing dial pers.
• The parameter cor-list-name is the COR list name.
• The parameter cor-list-number is a COR list identifier. The maximum number ofCOR lists
thatcan be created is 20.and the listsconsist of incoming or outgoing dial peers. The first
six COR lists are applied to a range ofdirectory numbers, 'fhedirectory numbers that do
not ha\e a COR configuration arc assigned lo the default COR list, as long asa default
COR list has been defined.
• The parameters starting-number - ending-number define the director- number range—for
example. 2000 to 2025.
• fhe ke\ word default instructs the routerto use an existingdefault COR list.
router(config-volcaport)#
translation-profile {incoming | outgoing} name
* Assigns a translation profile to a voice port
router(config-cm-fallback>#
I translation-profile {incoming | outgoing} name
' Assigns a translation profile to the call-manager-fallback
Cisco IOS service
The voice translation profiles can also be bound to call-manager-fallback Cisco IOS service.
The structure of the command is identical.
Note The incoming direction ofthe voice translation profile that is boundto the CiscoCallManager
fallback Cisco IOS service processes the calls comingfrom IP phones that are registered
with the router.
For more information about voice translation profiles, refer to Cisco TechNotes Number
Translation Using I oice Translation Profilesat
http:/^vww,cisco.coni/en/[;S/tech/tk652/tk90/technologies^ci)nfiguration_examplc()9l86a()08().5
fS18a.shtml and TechNotes Voice Translation Rules at
http://wwu.cisco.com/en/L;S/tech/tk652/lk90/iechnologies_tech..nole09186a0080325e8c.shlm!.
2-64 Implementing CiscoUnified Communications Manager, Part 2 (CIPT2) v8.0 12010 Cisco Systems. Inc
Cisco Unified SRST Dial Plan Commands: Number Modification
(Voice Translation Rules)
In voice translation rules, sets of number modification rules are defined.
router (confi gl #
voice-translation-rule number
router(cfq-translation-rulel#
rule precedence /match-pattern/ /replace-pattern/
[type {match-type replace-type} [plan {match-type replace-
type}]]
* Defines a translation rule
To define a translation rule for voice calls, use the voice-translalion-rulc command in global
configuration mode.
• Number: The number that identifies the translation rule. The range is from 1 lo
2147483647.
To define a translation rule, use the rule command in voice translation-rule configuration
mode.
• The parameter precedence defines thepriority of the translation rule, fhe range is from I to
15.
• The parameter Imatch-patternl is a stream editor (SED) expression thatis used to match
incoming call information, fhe slash (/) is a delimiter in thepattern.
• The parameter Ireplace-patternl isa SED expression that isused toreplace the match
pattern in the callinfonnation. Theslash is a delimiter in the pattern.
• The optional construct type match-type replace-type allows for modification ofthe number
type ofthe call. Valid values for the match-type argument are these: abbreviated, any,
international, national, network, reserved, subscriber, unknown. Valid values for the
replace-type argument are abbreviated, international, national, network, reserved,
subscriber, unknown.
• The optional construct plan match-type replace-type allows for modification ofthe
numbering plan of the call. Valid values for the match-type argument are any, data, ermes.
isdn, national, private, resen-ed, telex, unknown. Valid values for the replace-type
argument are data, ermes. isdn. national, private, reserved, telex, unknown.
router(config)#
I voice-translation-profile name
Defines a translation profile for voice calls
router(cfq-translation-profile)#
translate {called | calling | redirect-called | redirect-
target | callback} translation-rule-number
To define a translation profile for voice calls, you use the voice-translation-profile command
in global configuration mode.
The parametername definesthe name of the translation profile. The maximum lengthof the
voice translation profile name is 31 alphanumeric characters.
To associate a translation rule with a voice translation profile, you use the translate command
in voice translation-profile configuration mode:
• called: Associates the translation rule with called numbers
• redirect-target: Associates the translation rule with transfer-to numbers and call-
forwarding final destination numbers
• callback: Associates the translation rule with the numberto be used by IP phones for
callbacks
Note While ona call, IPphones display the calling-party number. When callbacks are placed from
call lists, the callback number {if present) is utilized for the outbound call, and notthe calling-
party number that was shown while the call was active.
• translation-rule-number: The number of the translation rule to use for the call translation.
The valid range is from 1 to 2147483647. There is no default value.
2-62 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) u8.0 >2010 Cisco Systems, Inc.
• The keyword extension-length sets the number of extension digits that will appear as a
caller ID followed by the parameter length, which is the numberof extension digits. The
extension length must match the setting for IP phones in Cisco Unified Communications
Manager mode, 'fhe range is from I to 32.
• The optional keyword evtension-pattern sets the leadingdigit patternof an extension
number when the patternis difTerent from the leadingdigits that are defined in the pat.jm
variableof the F.164 telephone number, such as when site codesare used. The parameter
extension-pattern that follows defines the leading digit patternof the extension number. It
comprises one or more digits and wildcard markers or dots (.). For example, 5.. would
include extensions 500 to 599: and 5... would include extensions 5000 to 5999. The
extension pattern configuration should matchthe mappingof internal to external numbers
in Cisco Unified Communications Manager.
• Theoptional keyword no-reg prevents the I7..164 numbers in the dial peerfrom registering
with the gatekeeper.
fhe example for the dialplan-pattcrn command shows how lo create a dial plan pattern for
directory numbers 500 lo 599that is mapped lo a DID range of 408 555-5000 to 5099. Ifthe
router receives an inbound call to 408 555-5044, then the dial plan pattern command is matched
and the extension of the called I7,. 164 number. 408 555-5000, is changed lo directory number
544. If an outbound calling-party extension number (544) matches thedial plan pattern, the
calling-part) extension will beconverted to the appropriate L.I64 number (408 555-5044). The
F..I64 calling-party number will appear as the caller ID:
Router (config)# call-manager-fallback
Router(config-cm-fallback)# dialplan-pattern 1 40855550..
extension-length 3 extension-pattern 5..
Since Cisco Unified SRST 8.0.the dialplan-pattcrn command has been used in the opposite
way with the addition of the keyword demote tothe end ofthe command. In this case it
demotes IF phone directory numbers thatarespecified in F.I64 format with a + prefix to
shorter extensions, which are to be used internally. Extemal callers place calls to the phones,
using E.164 format with a-f prefix. Ifthe calls are not natively received in this format from the
PSTN (which they rarely are), you have to transform the called number accordingly. Internal
users.howe\er. can dial each other by usingshorterextensions, which arc set up by the
dialplan-pattern command with the demote argument:
Router(config)# call-manager-fallback
Router(config-cm-fallback)# dialplan-pattern 1 +415526....
extension-length 5 demote
In this example, phones are configured with directory numbers+415526.... and have tobe
called that way from the outside. Internal users, however, can call each other by using the last
five digits (6....).
Note Thedialplan pattern command with thedemote argument isalso available in Cisco Unified
Communications Manager Expressand hence can also be used forCisco Unified SRST
when Cisco Unified Communications Manager Express is used in SRST mode.
router(conflq-if)#
isdn overlap-receiving [T302 ms]
router(conflq-cm-fallback)#
timeouts interdigit aec
router(config-cm-fallback)#
dialplan-pattern tag pattern extension-length length
[extension-pattern extension-pattern]
[no-reg] [demote]
The isdn overlap-receiving command is applicable on BRl interfaces or on the ISDN interface
ofTl/El controllers in PRI mode.
The optional parameter T302 defines the number of milliseconds that the T302 timer should
wait beforeexpiring. Validvalues for the milliseconds argumentrange from 500 to 20000.The
default value is 10000 (lOseconds).
To configure the timeout value to waitbetween dialed digits for all Cisco IP phones that are
attached to a router, usethe timeouts interdigit command in call-manager-fallback
configuration mode.
• The parameter sec defines the interdigit timeout duration, in seconds, for all Cisco IP
phones. Valid entries are integers from 2 to 120.
To create a global prefix that can be used to expand the extension numbers of inbound and
outbound calls into fully qualified E.164 numbers, youuse thedialplan-pattern command in
call-manager-fallback configuration mode.
• The parameter tag is the unique identifier that is used before the telephone number. The tag
number is any number from I to 5.
• The parameter pattern is the dial plan pattern, such as the area code, theprefix, and the first
oneor twodigitsof theextension number, plus wildcard markers or dots (.) for the
remainder of the extension number digits.
2-60 Implementing CiscoUnified Communications Manager, Pari2 (CIPT2) v8.0 ) 2010 Cisco Systems, Inc.
dial-peer voice 5 pots
description PSTN-emergency
destination-pattern 911
port 0/1/0
forward-digits all
Call Type
Emergency 911
Services [2-8)11
Local [2~9]xx-xxxx
Long distance or national 1[2-9]xx [2-9]xx-xxxx
International 011+countrycode+number
Toll-free 1[800,866,877,8 88]xxx-xxxx
Premium 1 900xxx-xxxx
976-xxxx
The table represents the most common classes ofPSTN calls in the Nortli American
Numbering Plan (NANP) and lists the pattern that is used for each class.
An access codeof 9 should be usedto indicate a PSTN call.
The patterns in the table are the minimum patterns that must be reachable in SRST mode. This *^**
example lists the configuration of dial peers that would be needed to reach all the numbers that
are indicated.
dial-peer voice 1 pots
description PSTN-LD
destination-pattern 91 [2-9] .. [2-9]
port 0/1/0
forward-digits 11
2-58 Implementing CiscoUnified Communications Manager, Part 2 (CIPT2) v8.0 >2010 Cisco Systems, Inc.
Note Details about the meanings of these special characters and about Cisco IOS dial peer
Ju configuration in general are provided in theImplementing Cisco Voice Communications and
QoS (CVOICE) course.
• The optional control character T indicates that the dcslination-pattcrn value is avariable-
length dial string. Using this control character enables the router to wait until all digits arc
received before routing the call.
To associate adial peer with aspecific voice port, use the port command in dial peer
configuration mode.
• The parameter slot-number defines the number of iheslot in the router inwhich the voice
interface card (VIC) is installed. Valid entries depend onthe number ofslots that the router
platform has.
• The parameter/jo/7 defines the voice port number. Validentries are 0 and I.
mm
m»
router(config)M
| dial-peer voice tag [pots | voipj
- Defines the dial peer
router(config-dial-peer) #
|destination-pattern [+]Btring[T]
3
Specifies either the prefix or the complete E.164 telephone
numbertobeused tbradial peer
router(conflg-dlal-peer)B
I port slot-number/port
• Associates a dial peer with a specific voice port
To define a particular dial peer, specify a voice encapsulation method, and enter dial peer
configuration mode, you use the dial-peer voice command in global configuration mode.
• The parameter tag specifies digits that define a particular dial peer. The range is from 1to
2147483647.
• The keyword pots indicates that this peer isa plain old telephone service (POTS) peer;
voip indicates that this peer is a VoIPpeer.
To specif; either the prelix orthe complete E. 164 telephone number tobe used for a dial peer,
you use the destination-pattern command in dial peer configuration mode.
• The optional character+ indicates that an E.164 standard numberfollows.
• The parameter string defines aseries ofdigits that specify apattern for the E. 164 orprivate
dialing plan telephone number. Valid entries are the digits 0 through 9, the letters A
through D. and these special characters:
2-56 implementing Cisco Unified Communications Manager, Part 2 (CIPT2) vS.O >2010 Cisco Systems, Inc.
Additional Cisco Unified SRST Dial Plar
Requirements (Cont.)
Ifthe calling privileges (which in normal mode are controlled by Cisco Unified
Communications Manager) have tobe preserved in SRS'f mode, class ofrestriction (COR)
configuration has to be used.
fhe handling ofvariable-length numbers should also be preserved in SRST mode, fhis
includes tuning ofthe interdigit timeout, the possibility to use the # key toterminate dialing:
and the implementation of overlapsending.
Ideally, the numbers in call lists (such as missed calls) have the correct format (PSI'N access
code plus PSTN phone number) that is required for callback so that users do not have to edit
the number manually. In this case, the calling party ID of incoming calls from the PSTN needs
to be modified by voice translation profiles and voice translation rules.
Abbreviated dialing between sites of the site code plus the extension number is possible in
SRST mode. Voice translation profiles have to be used to expand the called numbers to PSTN
format for intersite dialing.
2-54 ImplementingCisco Unified Communications Manager, Part 2 (CIPT2)v8.0 >2010 Cisco Systems, Inc.
Cisco IOS Gateway MGCP Fallback and Cisco
Unified SRST Dial Plan Configuration
This topic describes the minimum dial pian configuration steps that are needed for
communication between phones in SRST mode and the PSTN.
The minimum requirement for a dial plan in SRST mode is that it must enable the remote site
users to place and receive calls from the PSTN.
At least one dial peer must be configured lo enable calls to the PSTN. The destination pattern
of that dial peer has to correspond to the PS IN access code (for example, 9T). 'fhe more
elegant wa\ is to configure several dedicated dial peers with destination patterns that match the
number patterns in a closed numbering plan, such as 91 (91 followed by 10 dots).
In countries that have open numbering plans, the only destination pattern that is needed is 9T.
Because of the variablelength of dialednumbers, the router is waitingfor the interdigittimeout
(1302) or for a hash (#) sign to indicate the end of the dial string.Cisco Unified SRSTversion
4.1 and Cisco Unified Communications Manager F.xpress Version 4.1 do not support the
overlap sending feature to the PSI'N. fhe receiving of ISDN overlap dialing from PSTN is
supported but has lo be enabledon the interlaces. To shorten the wail time for users afterthe;
complete the dial string, it is possible to reduce the inlerdigit timeout from thedefault of 15
seconds.
Dial plan pattern configuration is a powerful tool forthe modification of incoming called
numbers to match remote site extensions.
2-52 Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 ©2010 Cisco Systems, Inc
Step 3: Configuring CFUR
"fheCFUR feature is a way to reroutecalls placed to a temporarily unregistered destination
phone. Theconfiguration of CFUR consists of twomain elements: destination selection and
CSS.
When the directory number isunregistered, calls can be rerouted to the voice mail that is
associated with theextension or to a directory number that is used to reach the phone through
the PSTN. The latter approach ispreferable when a phone islocated within a site whose VAN
link is down. Ifthe site is equipped with SRST. thephone (and itseolocated PS'fN gateway)
will reregister with the eolocated SRST router. The phone isthen able to receive calls placed to
its PSI'N direct inward dialing (DID) number.
In this case, the appropriate CFUR destination isthe corresponding PSTN DID number ofthe
original destination directory number. Configure this PSTN DID in the destination field, along
with applicable access codes and prefixes (for example. 9 1415 555-1234).
Cisco Unified Communications Manager attempts toroute the call tothe configured destination
number b\ using the CFUR CSS ofthe called directory number. The CFUR CSS is configured
on the target phone and is used by all devices that are calling the unregistered phone.
As a result, all calling devices will use the same route pattern, route list, and route group to
place the call. Ifaspecific route group is defined in the route list, all CFUR calls to agiven
unregistered device will be routed through the same unique gateway, regardless ofthe location
ofthe calling phone. In this case, it is often recommended that you select a centralized gateway
as the egress point to the PSTN for CFUR calls and configure the CFUR CSS to route calls to
the CFUR destination through this centralized gateway.
Abetter solution istouse the local route group feature. When you use this feature, the route list
does not refer to a specific route group, but "Standard Local Route Group" is added to the route
list instead, fhe route group that isto be used for the calls is then determined by the local route
group that is configured at the device pool ofthe calling device. In this case, phones at different
sitescan refer to different route groups viatheirdevice pool configuration.
; r,-.^l!J-i-E'™' ,; '-1
,F4jsf
' 1-ti. - 1 » • -" - 1•'* '•.' or\i VI'' me liriTdr^eriiai' v^ Onlyafter the FtrBl ihueiMicn
jO^LC^-^i^_^ _ ^ o
This parameter specifics the maximum number offorward unregistered hops that are allowed
for a director, numberat one time. It limitsthe numberof times that the call can be forwarded
because of the unregistered directory number when a forwarding loop occurs. Use this count to
stop forward loops for external calls that have been forwarded by CFUR, such asintercluster IP
phone calls and IP phonc-to-PSTN phone calls that are forwarded toeach other. Cisco Unified
Communications Manager terminates thecall when the value that is specified in thisparameter
is exceeded, fhe default 0 disables the counter but not the CFUR feature. The allowed range is
from 0 to 60.
2-50 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 ©2010 Cisco Systems, Inc
Cisco Unified Communications Manager Dial
Plan Configuration for SRST Support
This topic describes theconfiguration that is necessary for adjusting the Cisco Unified
Communications Manager dial plan to work with Cisco Unilied SRST.
TheCisco Unified SRST router is installed at a small branch officesitewith three IP phones.
Here is the configuration that is necessary forthe Cisco Unified SRST routerto perform the
MGCP gateway fallback in this environment:
SRST# configure terminal
SRST(config)#ccm-manager fallback-mgcp
SRST(config)#application
SRST(config-app)#global
SRST(config-app-global)#service alternate Default
SRST(config-app-global)#end
SRST#
2-48 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 ) 2010 Cisco Systems, Inc
Steps 1 and 2: Enabling MGCP Fallback and Setting Fallback
Service
The Cisco IOS command ccm-manager fallback-mgcp enables the gateway fallback feature
and allows an MGCP voice gateway to provide call-processing services through Cisco Unified
SRST or other configured applications when Cisco Unified Communications Manager is
unavailable.
router (config)#
ccm-manager fallback-mgcp
or
router Iconfig-app-global)W
service alternate Default
The call application alternate Default command specifies that the default voice application
takes o\er ifthe MGCP call agent is not available, fhiscommand allows a fallback to H.323 or
SIP. which means that local dial peers are considered for call routing.
You enter the sen ice alternate Default command in the global application configuration
mode. To na\igate to this location, perfonn these steps:
Step 1 To enterapplication configuration mode to configure applications, use the
application command in global configuration mode.
Step 2 To enter global application configuration mode, use the global command in
application configuration mode.
Enter eitherof the two commands, depending on the Cisco IOSSoftware release. The newer
configuration method is the servicecommand.
As discussed in the previous lesson, analog calls are preserved in the event of MGCP fallback.
In order toprovide call preservation during switchback, you must enable call preservation for
H.323 using the following commands:
voice service voip
h323
Configuration of the MGCP gateway fallback on a Cisco IOS router to support the MGCP
fallback function requires these two steps:
• Activation of MGCP gateway fallback
• Definition of the service to fall back to
Toenable outbound calls while in SRST mode on an MGCP gateway, youmust configure two
fallback commands on the MGCP gateway. These two commands allow SRST to assume
control overthevoice portandovercall processing on the MGCP gateway. With CiscoIOS
Software releases earlier than 12.3(14)T, configuration of MGCP gateway fallback requires the
use of the ccm-manager fallback-mgcp andcall application alternatecommands. With
Cisco IOS Software releases later than I2.3(I4)T, configuration of MGCP gateway fallback
requires the use of the ccm-manager fallback-mgcp and service commands.
Note Both commands have to beconfigured. Configurations will not be reliable if only theccm-
manager fallback-mgcp command is configured.
2-46 Implementing Cisco Unified Communications Manager. Part2 (CIPT2) v8.0 >20IOCisco Systems, Inc.
Cisco Unified SRST Configuration Example
fhe figure shows anexample of a Cisco Unified SRST configuration to support SCCP-
controlled IP phones.
"fhe SRST router is installed at a small branch office site with three IP phones, each having two
lines (six lines intotal). The IP address 172.47.2.1 isconfigured onthe Fthemct interface where
the IP phones are connected. Here isthe configuration thai you must perform for the Cisco
Unified SRST router to operate in this environment:
SRSTtt configure terminal
SRST(config)# call-manager-fallback
SRST(config-cm-fallback)# ip source-address 172.47.2.1 port
2C00
Note More commands might be necessary, depending on the complexity ofthe deployment
router(config-cm-fallback)#
|limic-dn (7910 / 7335 / 7940 / 7960} max-lines
* Limitsthe directory number lines on Cisco IP phones during
SRST mode.
* Default is 6-line maximum.
router(config-cm-fallback} #
Ikeepalive oecondg
* Sets the time interval, in seconds, between keepalive
messages that are sent to the router by Cisco IP phones.
• Default is 30.
The optional Cisco IOS command limit-dn limitsthe directory number lineson Cisco IP
phones during SRST mode, depending on device types.
Note This command mustbe configured during initial Cisco Unified SRST router configuration,
before any phone actually registers with the Cisco Unified SRST router. However, the
number of lines can be modified at a later time.
The setting for maximum lines is from 1to 6.The default number of maximum directory lines
is setto 6. Ifthere is any active phone with the lastline number greater than this limit, warning
infonnation is displayed for phone reset.
The optional Cisco IOS command keepalive sets the time interval, in seconds, between
keepalive messages thataresent to therouter by Cisco IPphones. Therange is 10 to 65535.
Default is 30.
The keepalive interval is the lime between keepalive messages that aresentby a network
device.
2-44 Implementing Cisco Unified Communicalions Manager, Part2 {CIPT2) v8.0 12010 Cisco Systems, Inc.
Steps 3 and 4: Setting Maximum Directory Numbers and
Telephones
"fhe ne\t two commands, max-dn and max-ephone. are mandatory because the default value?
for both commands are defined as 0.
rouCer(conflg-cm-fallback)W
max-dn max-directory-numbers [dual-line] [preference
preference-order]
• Sets the maximum number of directory numbers or virtual
voice ports that can be supported by the Cisco Unified SRST
router and activates the dual-line mode
The Cisco IOS command max-dn sets the maximum numberof directory numbers or virtual
voice ports that can be supported by the router, and activates the dual-line mode. The maximum
numberis platform-dependent. The default is 0.
The dual-line keyword isoptional. Itallows IP phones in SRS'f mode tohave a virtual voice
port with two channels.
Note Thedual-line keyword facilitates call waiting. Call Transfer, and conference functions by
allowing two calls tooccur onone line simultaneously. In dual-line mode, all IP phones on
the Cisco Unified SRST router support two channels per virtual voice port.
The optional parameter preference sets the global preference for creating the dial peers for all
director) numbers that are associated with the primary number. The range is from 0to 10. The
default is 0. which is the highest preference.
Note The router must berebooted in order to reduce thelimit ofthedirectory numbers orvirtual
voice ports after the maximum allowable number is configured,
To configure the maximum number ofCisco IP phones that can be supported by aCisco
Unified SRS'f router, use the max-ephones command incall-manager-fallback configuration
mode. The default is 0. and the maximum configurable number is platform-dependent.
Note The router must be rebooted in order to reduce thelimit ofCisco IP phones after the
maximum allowable number is configured.
router (confiq)#
call-manager-fallback
router (config-cm-fallback)#
ip source-address ip-address [port port]
[any-match strict-match]
The Cisco IOS command ip source-address enables therouter to receive messages from the
Cisco IP phones through the specified IP addresses and provides for strictIP address
verification. The default port number is 2000. This IPaddress will besupplied later asan SRST
reference IP address in Cisco Unified Communications ManagerAdministration.
The ip source-address command is a mandatory command. Thefallback subsystem does not
start if the IP address ofthe Ethernet port towhich the IPphones are connected (typically the
Ethemet interface of the local Cisco Unified SRST gateway) isnot provided. Ifthe port number
is not provided, the default value (2000) is used.
The any-match keyword should beused to instruct the router topermit Cisco IP phone
registration even when the IP server address that is used by the phone doesnot match the IP
source address. You can use this option to allow registration of Cisco IPphones on different
subnetsor on subnetswithdifferent default DHCProutersor different TFTP server addresses.
The strict-match keyword should beused to instruct the router to reject Cisco IPphone
registration attempts ifthe IP server address that isused by the phone does not exactly match
the source address. By dividing the Cisco IP phones into groups on different subnets and giving
each group difTerent DHCP default-router orTFTP server addresses, this option can beused to
restrict thenumber of Cisco IP phones that areallowed to register.
2-42 Implementing Cisco Unified Communications Manager. Part 2 (CIPT2) v8.0 >2010 Cisco Systems, Jnc.
Cisco IOS Gateway SRST Configuration
This topic describes the configuration steps for enablingCisco Unified SRSTon the Cisco IOS
router.
To configure Cisco Unified SRST on a Cisco IOS router tosupport the Cisco IP phone
functions, follow these steps:
Step 1 Fnable Cisco Unified SRST. sothat itenters call-manager-fallback configuration
mode.
Step 2 Define the IP address and port to which the SRS'f service binds.
Step3 Define the maximum number of directory numbers to support.
Step 4 Define the maximum number of IP phones to support.
Step 5 Define the maximum numbers thatare allowed per phone tjpe.
Step 6 Definethe phone keepalive interval.
Tip When Cisco Unified SRST is enabled, Cisco IP phones do not have to be reconfigured while
in catl-manager-fallback configuration mode, because phones retain the same configuration
that was used with Cisco Unified Communications Manager.
Locator
Nefwort Locale
BkST Reference*
Physical Location
Administrators select the configured SRST reference from the drop-down menu in the device
pool configuration.
Note Ifdevices are associated with this SRST reference, a message is displayed, saying that
devices must be reset for the update to take effect.
2-40 implemenbng Cisco Unified Communications Manager, Part 2 (CJPT2) v8.0 >201DCisco Systems, inc.
Step 1: SRST Reference
An SRST reference comprises the gateway, which can provide limited Cisco Unified
Communications Manager functionality when all other Cisco Unilied Communications
Manager servers for an IP phone are unreachable.
mm
IF i^ddr-5G" ]•;;•;!
SIP NtVAC'k IP ASd-csi |
SRST references detemiine the gateways that IP phones will search when (hey attempt to
complete a call if the Cisco Unified Communications Manager is unavailable.
Administrators must configure Cisco Unified Communications Manager with a uniqueSRST
reference name that specifies the IP address of (he CiscoUnilied SRSf gateway. The default
TCP port number 2000 is normally used.
The SIP network and IP address applies to Cisco Unified SIP SRST. If Cisco Unified SIP
SRST is used, the IP address and port that are used by the SIP protocol of the Cisco Unified
SRSf gateway have lo be specified: thedefault port number is 5060. Theconfigured address
and port will be used by SIP phones to register with theCisco Unified SIP SRST gateway.
For Cisco Unified SRSTgateways that supportSCCPphones with defaultport number2000
with secure SRST disabled, it is not necessary to add an SRST rclcrcncc if the IP address of the
Cisco Unified SRST gateway is the default gateway of the IP phone. In this case,you can use
the option Use Default Gateway at the device pool of the affected IPphones.
The role of Cisco Unified Communications Manager regarding the SRST feature is to provide
the phones with the needed information for finding the relevant SRST gateway to register with
when they lose contact with Cisco Unified Communications Manager subscribers.
The first step is to define an SRST reference. This reference contains information about IP
addresses and ports of SRST gateways for SCCP and Session Initiation Protocol (SIP) phones.
Because the SRST functions are different for SIP and for SCCP, the addresses and ports are
also different.
The second step is to provide a group of phones with this information by assigning the SRST
reference to a proper device pool, which is then assigned to the phones.
2-38 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 >2010 Cisco Systems, Inc.
MGCP Fallback and Cisco Unified SRST Configuration
Requirements
When configuring MGCP fallback and Cisco Unified SRST,you must take several steps at
different locations.
You need to use the Cisco Unified Communications Manager Administration to define the
SRST references for phones. You must also configure (he Call Forward Unregistered (CFUR)
feature and set the CFUR destination of lines on remote site phones to the correct public
switched telephone network (PSTN) number on the Cisco Unified Communications Manager
Administration to enable reachable remote sites in SRS'f mode.
On Cisco IOS gateways, you must enable and configure the MGCP fallback and Cisco Unified
SRSf features. In addition, you must implement a simplified SRST dial plan on the remote site
gateways to ensure connectivity for remote sites in SRST mode.
Cisco Unified
Communications
Manager
In Cisco IOS
Software, MGCP
fallback and Crsco
Unified SRST must
be enabled and
configured.
Activate and configure the MGCP gateway fallback feature on the Cisco IOS router.
Cisco Unified SRST must be configured on the side of the Cisco Unified Communications
Manager and on the side of the Cisco IOS router.
2-36 Implementing Cisco Unified Communications Manager, Part2 (CIPT2) v8.0 >2010 Cisco Systems, Inc.
Lesson 2
-**»
Objectives
Upon completing this lesson. \ouwill be able to configure Cisco Unified SRST to provide call
survivability for IPphones, and MGCP fallback for gateway survivability, fhis ability includes
m+
beingable to meet theseobjectives:
• Describe the configuration requirements at Cisco Unified Communications Manager and at
the SRST gateway
• Describe how lo configure Cisco Unified Communicalions Manager to enable SRSI for
remote phones
• Describe how to configure a Cisco IOS router for SRS'f
a Describe how to configure a Cisco IOS router to support MGCP fallback
• Describe how to configure Cisco Unified Communications Manager to route calls to
unregistered devices via the PSTN
• Describe how to configure a Cisco IOS router with a dial plan for SRST operation
•mm
Summary
This topic summarizes the key points that were discussed in this lesson.
Summary
References
For additional information, refer to these resources:
2-34 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 © 2010 Cisco Systems, Inc.
Cisco Unified SRST Dial Plan Requirements Example
Call-routing components on Cisco IOS routers and Cisco Unified Communications Manager
arc necessary before a dial plan will work in SRS'f mode.
Configuration includes
dial plan pattern and voice
translation profiles to allow
extension-only dialing.
CFUR must be defined on the Cisco Unified Communications Manager side. Configuring the
Cisco IOS router is a little more complex when vou use dial peers, COR. dial plan pattern, and
\oice translation profiles to define the simplified Cisco Unified SRST dial plan.
However, when IP WAN connectivity is lost betweena branch site and the central site, Cisco
Unified SRSTtakes controlof the branch IP phones, and the entire configuration that is related
to partitions andCSSs is unavailable until IP WAN connectivity is restored. Therefore, it is
desirable to implement classes of service within the branch router when running in SRST mode.
Forthis application, you mustdefine classes of service in Cisco IOSrouters by using the class
of restriction (COR) functionality. You can adapt the COR functionality to replicate the Cisco
Unified Communications Manager concepts of partitionsand CSSs by following these main
guidelines:
• Define named tags for each type of call that you want to distinguish.
• Assign basic outgoing COR listscontaining a single tag each to the outgoing dial peersthat
should not be available to all users. These outgoing COR lists are equivalent to partitions in
Cisco Unified Communications Manager.
• Assigncomplexincoming COR lists containing one or more tags to the directory numbers
that belong to the various classes of service.
2-32 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v9.Q >2010 Cisco Systems, Inc.
CFUR Example with Globalized Call Routing
The figure shows the same scenario, but thistimewith globalized call routing.
Route Pattern
Partition. System
Device Pool
LRG Site2-GW
3001
Phone
CFUR *152155530O1
CFUR CSS System
There is only a single \+! route pattern, the referenced route list has local route groups enabled.
All phones use the same CFUR CSS. which provides access to the partition of the global route
pattern, fhe egress gatewa; is selected by the local route group feature. Localization of the
called number occurs at the egress gateway by global transformations.
Ifa called is placed to an unregistered phone of Site l, die CFUR destination+152I5553001 is
called using the single off-net route pattern, which is configured to use the local route group (in
the referenced route list). Consequently, like with any other PSTN call, CFUR calls use the
local gateway instead of the HQ gateway, regardless of the location of the caller. There is no
need for all callers to use the same gateway for CFUR calls. In addition, all CFUR destination
numbers are specified in global format (E. 164 with +- prefix).
Ptione
CFUR 915215553001
JP ;.,)!: K)ir> hiic;2 ph:*^; tc '~Wt ' pSn">si<; t;;K-.s HG
CFUR CSS HQ
There are three sites: HQ, Site I, and Site 2. The remotesites are backed up by SRS'f gateways.
If IP connectivity between site 1 andthe HQfails, Site 1 phones will failover to SRST mode.
They can still call the HQ and Site 2 viathe PSTN. When an HQ phone attempts to call a phone
at Site 1—which is unregistered in Cisco Unified Communications Manager—the call is placed
to the CFUR destination configured at the Site 1phone (915215553001 in thisexample). The
CFUR CSS of the Site 1 phoneensuresthat a route pattern9.@—which refers to the HQ
gateway—can be accessed. Therefore, the call is redirected to the PSTN number of the called
phone and sent to the HQ gateway.
When a user at Site 2 attempts to call a phone at Site 1, the same thing happens. The CFUR
destination 915215553001 is called using the CFUR CSS configured at the Site 1 phone and
therefore matches the 9.@ route pattern that is referring to the HQgateway and not to a 9.fy
route pattern referring to a Site2 gateway. Therefore, the call will utilize the IP WAN to get
from Site 2 to the HQ and from there it will break out to the PSTN towardsSite 1.
If there were more sites, they would all use the HQ gateway for CFUR calls to Site 1. This can
lead to suboptimal routing. In addition, differentroute patterns may be needed depending on
the destination of the CFUR call. In an international deployments, the CFUR destination
numbermay be a mix of national and international numbers. Each destination numberhas to be
specified in a way that it can be routed by the CFUR CSS. There is no common format for all
CFUR destinations—some may be specified in national format, others in international format.
2-30 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 © 2010 Cisco Systems, Inc.
CFUR Interaction with Globalized Call
Routing (Cont.)
Without using local routegroups, the CFUR CSS determines the gateway that is used for the
CFUR call. The CFUR CSS of the phonethat is unregistered is used not the one of the phone
that tries to reach the unregistered phone. This means that all callers use the same CFUR CSS
when calling an unregistered phone (the CFUR CSS configured at the destination phone).
Consequently, if callers arc located at different sites, they will all use the same gateway for the
CFUR call. Usually the main site gateway is used for that purpose; that means that the CFUR
CSS (applied to all phones) provides access lo PSTN route patterns that use the main site
gateway (via the referenced route list and route group).
With local route groups, each caller can use its local gateway for CFUR calls; there is no need
to use the IP WAN touards the main site and then break out to the PS'FN with the CFUR call at
the main site gateway. Depending on the deployment this can be a huge improvement for
reaching sites that lost IP connectivity to Cisco Unified Communications Manager.
2-28 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 >2010 Cisco Systems, Inc.
a director;' numberwhose CFUR setting is configured for voice mail. At the same time, this
configuration would alsolimit potential loops to two fordirectory numbers whose CFUR
configuration sends calls through the PSTN.
Note Cisco Unified Communications Manager Extension Mobility directory numbers should not be
configured to send CFUR calls to the PSTN DID that is associated withthe directory
number. The directory numbers of Cisco Unified Communications Manager Extension
Mobility profiles in the logged-out state are deemed to be unregistered; therefore, any calls
to the PSTN DID number of a logged-out directory number would trigger a routing loop To
ensure that calls made to Cisco Unified Communications Manager Extension Mobility
directory numbers in the logged-out state are sent to voice mail, you must configure their
corresponding CFUR parameters to send calls to voice mail.
CFUR Considerations
As mentioned earlier, the CFUR feature allows calls that are placed to a temporarily
unregistered phone to be rerouted to a configurable number.The configuration of CFUR
comprises two main elements;
• Destination selection: When the directory number is unregistered, calls can be rerouted to
voice mail or to the directory numberthat was used to reach the phone through the PSTN.
• Calling search space (CSS): Cisco Unified Communications Manager attempts to route
the call to the configured destination number using the CFUR CSS of the directory number
that was called. The CFUR CSS is configured on the target phone and is used by all devices
that are calling the unregistered phone.
If a phone is unregistered while the gateway that is associated with the direct inward dialing
(DID) number of that phone is still under the control of Cisco Unified Communications
Manager, CFUR functionality can result in telephony routing loops. For example, if a phone is
simply disconnected from the network, the initial call to the phone would prompt the system to
attempt a CFUR call to the DID of the phone through the PSTN. The resulting incoming PSTN
call would, in turn, trigger another CFUR attempt to reach the directory number of the same
phone, triggering yet another CFUR call from the central PSTN gateway through the PSTN.
This cycle could repeat i'sclf until system resources are exhausted.
'fhe Cisco CallManager service parameter Max Forward UnRegistered Hops to DN in the
Clusterwide Parameters (Feature—Forward) section in Cisco Unified Communications
Manager Administration controls the maximum number of CFUR calls that are allowed for a
directory number at one time. The default value of 0 means that the counter is disabled. If any
directory numbers are configured to reroute CFUR calls through the PSTN, loop prevention is
required. Configuring this service parameter to a value of 1 would stop CFUR attempts as soon
as a single call is placed through ihe CFUR mechanism. This setting would also allow only one
call to be forwarded to voice mail, if CFUR is so configured. Configuring this service
parameter to a value of 2 would allow up to two simultaneous callers to reach the voice mail of
2-26 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 ) 2010 Cisco Systems, Inc.
Ensure Connectivity from Main Site Using CFUR
During fallback, main site users still should be able locall remote site users by using their
extension numbers.
Using CFUR
Communications
Manage
Cisco Unified Communications Manager considers the remote site phones as unregistered and
cannot route calls to the affected IP phone directorv' numbers. Therefore, if main site users dial
internal extensions during the IP WAN outage, the calls will fail (or go to voice mail).
To allow remote IP phones to be reached from other sites, you can configure Call Forward
Unregistered (CFUR) at the remote site phones. You should contigurc the CFUR destination at
each remote IP phone with the PSTN number of the IP phone so that internal calls from other
sites get forwarded to the PSTN number of an IP phone that is currently unregistered and is
therefore not reachable over the IP network.
©2010 Cisco Systems. Inc Centralized Cal I-Processing Redundancy Implementation 2-25
Ensure Connectivity for Remote Sites
When Cisco Unified SRST is active, you musttakeseveral measures to ensure connectivity
from remotesites to PSTN destinations, betweendifferentsites, and insidethe site itself.
PSTN connectivity:
* You must implement dial peers with destination patterns
corresponding to the PSTN access code.
• Voice translation profiles modify the calling-party number to
enable callback.
To guarantee PSTN connectivity, you must implement dial peers with destination patterns
corresponding to the PSTN access code. In H.323or SIP gateways, these dial peers must be
presentfor normal operation. When MGCP gateways are used, dial peers are activated by the
MGCP gateway fallback mechanism. Interdigit timeout adopts open numbering plans that do
not have a fixed number of digits.
Voicetranslation profilesthat are applied to dial peers, the voice interface, or the voice port
modify the calling party ID to enable callback from call lists.
For intrasite and intersite connectivity, voice translation profiles are configured to expand
called numbers to PSTN format during fallback.
The Cisco IOS command dialplan-pattern in call-manager-fallback configuration mode
modifies incoming called numbers to match the remote site extensions, 'fhis command also
ensures that internal extensions can be dialed even though the lines are configured with the site
code and extension, "fhe Line Text Label settings that are defined in Cisco Unified
Communications Manager will not be applied to the Cisco Unified SRST phones, so the
complete directory number that is applied to the line will be visible to the user.
2-24 Implementing Cisco Unified Communications Manager, Pan 2 (CIPT2) v8.0 12010 Cisco Systems, Inc.
Dial Plan Requirements for MGCP Fallback and
Cisco Unified SRST Scenarios
Thistopic describes therequirements of standalone dial plans for working with MGCP fallback
and Cisco Unified SRST.
Unified
Communicalions
Manager
SRST failo\er lea\es the remote site independent from the complex dial plan that is
implemented in Cisco Unified Communications Manager in the main site. The Cisco Unified
SRST router needs to have a minimal dial plan that is implemented lo allow for all remote site
phones, all main site phones, and all PSTN destinations to be reached with the same numbers as
in standard mode.
During fallback, users should be able to dial main site directorv' numbers as usual. Because
these calls have to be routed over the PSTN during fallback, main site extensions have to be
translated to E.164 PSTN numbers at the PSTN gateway.
Most enterprises limit the range of destinations that are reachable from specific extensions by
applying a class of service to the extensions, fhis limitation should still be valid during times in
SRST mode.
>2010 Cisco Systems, Inc. Centralized Cal I-Processing Redundancy Implementation 2-23
Support for Multiple MOH Sources
Cisco Unified SRST v8.x also introduces support for multiple music on hold (MOH) sources.
Before Cisco Unified SRST v8.x. only asingle MOH file was supported by Cisco Unified
SRST. Cisco Unified Communications Manager Express in SRST mode, and Cisco Unified
Cisco Communications Manager Express in standalone mode.
Cisco Unitied SRST v8.x allows you toconfigure up tofive additional MOH sources by
configuring MOH groups.
Only SCCP IP phones support these newly introduced MOH groups. You can configure each
MOH group with an individual MOH file that is located in the flash memory of the router, and
you can enable multicast MOH for each MOH group. Each MOH group is configured with the
directory' number ranges that should utilize the corresponding MOH group when callers are put
on hold.
The traditional MOH configuration for Cisco Unified SRST and Cisco Unified
Communications Manager Express is still supported. Itisused by all phones that do not have a
MOH group assigned. All of these phones are SIP and SCCP phones whose directory numbers
have not been specified in any MOH group.
MOH files can becached inrouter RAM. This process isuseful toreduce the amount of read
operations in flash, but it requires enough available RAM at the router. You can specify a
maximum size per MOH file in order to limit RAM usage for MOH file caching.
2-22 Implementing Cisco Unified Communications Manager, Part2 (CIPT2) v8.0 ) 2010 Cisco Systems. Inc.
Plus (+) Prefix and E.164 Support in Cisco Unified SRST
Cisco Unified SRST version 8.0 introduces support for directory numbers in E. 164 format with
a plus (-t-) prefix.
SIP and SCCP IP phones can fallback to SRST and register with adirectory number in E. 164
format with a+prefix. Assigning directory numbers in E. 164 format ensures globally unique
numbers: the + sign is prefixed in orderto indicate thatthe number is in E. ]64 format.
Cisco Unified SRST and Cisco Unified Communications Manager Express in SRST mode
allow internal callers to use internal extensions for calling IPphones that have numbers in
E.164 format. A new dial plan pattern command hasbeen introduced with Cisco Unified SRST
v8.x to achieve the demotion of the I:. 164 number to the internally used shorter numbers. While
the standard dial plan pattern command expands to a longer PSTN format any directory
numbers that are applied tophone lines, the new dial plan pattern command has the opposite
function: Itallows internal callers to dial shorter, internally used extensions, which are
expanded to the applied directory numbers in E.164 format.
Outside callers dial the IP phone directory numbers as configured—with a 4 prefix and the
complete E.164 number.
At the IPphones, the calling-party number that isshown on the phone display can be
transformed independently from the number that will be used for callback. This transformation
is possible because ofa newly introduced translation type in voice translation profiles—a
translation rule of the callback number.
Objectives
Uponcompleting this lesson,you will be able to implement a dial plan to supportinbound and
outbound PSTN dialing, site-code dialing, and TEHO in an international environment. This
ability includes being able to meet these objectives:
• List dial plan issues and possible solutions
• Describe how site codes and transformation masks solve issues that are caused by
overlapping directory numbers
• Describe how to implement PSTN access in a multisite deployment
• Describe how to implement selective PSTN breakout
• Describe how to use the PSTN as a backup for calls to other VoIP domains
• Describe how to implement TEHO
• Describe the concept of globalized call routing and how it simplifies dial plans in
international multisite deployments with centralized call processing
• Explain special considerations for implementing globalized call routing
Multisite Dial Plan Overview
This topic lists dial plan requirements for multisite deployments with centralized call
processing.
In multisite environments with centralized call processing, you use these dial plan solutions:
• Access and site codes: By adding an access code and a site code lo director},' numbers of
remote locations, you can provide call routing that is based on the site code instead of on
directory numbers. As a result, directory numbers do nol have to be globally unique,
although they must be unique within a site. Configuration requires route patterns,
translation patterns, partitions, and calling search spaces (CSSs).
• Implementing PSTN access: You implement PSTN access within a Cisco Unified
Communications Manager cluster by using route patterns, route lists, route groups,
partitions, and CSSs. When implementing TEHO. you use the same dial plan configuration
elements: however. \ou have lo configure more entities, which makes the configuration
more complex.
• Implementing PSTN backup: The IP WAN that is used in a multisite deployment with
centralized call processing is backed up by Media Gateway Control Protocol (MGCP)
fallback. Cisco Unified Survivable Remote Site Telephony (Cisco Unified SRST) or Cisco
Unified Communications Manager Express in SRS'f mode, and Call Forward
Unregistered(CFUR).
1-108 ImplementingCisco Unified Communications Manager, Part 2 (CIPT2)v8 0 12010 Cisco Systems, Inc
Dial Plan Requirements for Multisite Deployments with
Distributed Call Processing
Dial plan requirements for multisite deployments with distributed call processing are like the
dial plan requirements ofmultisite deployments with centralized call processing.
In multisite environments with distributed call processing, you use these dial plan solutions:
• Access and site codes: Byadding an access codeanda site codeto directory numbers of
remote locations, you can provide call routing thatis based on thesite code instead of on
directory' numbers. As a result, directory numbers do nothave to be globally unique,
although they must be unique within a site. Configuration elements include route patterns
and translation patterns.
• Implementing PSTN access: You implement PSTN access within a Cisco Unified
Communications Manager cluster by usingroute patterns, route lists, routegroups,
partitions, and CSSs. When implementing TEHO, you use thesame dial plan configuration
elements; however, you have to configure more entities, which makes the configuration
more complex.
• ImplementingPSTN backup: Backup of the IP WAN is provided by route lists androute
groups with on-net (prioritized) and off-net (PSTN)paths.
aH Agent
1-110 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2] uB.O © 2010 Cisco Systems. Inc.
CCD solvesdial plan scalability issues by allowingCisco Unified Border Element, Cisco
Unified SRST.Cisco Unified Communications Manager, Cisco Unified Communications
Manager Express, andCisco IOS gateways to advertise and learn call-routing information in
the form of internal directory numbers and PSTN numbers or prefixes.
CCD utilizes the Cisco ServiceAdvertisement Framework (SAF).SAF is a network-based,
scalable, bandwidth-efficient, real-time approach to service advertisement and discovery.
mm
Cisco Unified
Communication;]
Manager
Nonconsecutive
£000-2157 Numbers
2365-2999 "♦"
Inthe example, two sites have overlapping and nonconsecutive directory' numbers. To
accommodate unique addressing of all endpoinls. site-code dialing is used. Users dial anaccess
code (8 in this example), followed by a three-digit site code. When calling thephone with
directory number 1001 at the remote site, a user who islocated atthe main site has to dial
8222.1001. For calls in the other direction, remote usersdial 8111-1001. Whendistributed call
processing is used, each Cisco Unified Communications Manager cluster isaware ofonly its
own director) numbers indetail. For all directory numbers that are located at the other site, the
call is routed to a Cisco linified Communications Manager server at the other site that is based
on the dialed site code.
1-112 Implementing Cisco Unified Communications Manager. Part 2 (CIPT2) v8.0 ) 2010 Cisco Systems. Inc
Digit Manipulation Requirements for Using Access and Site
Codes
The figure shows digit manipulation requirements for site code implementation.
Cisco Unified
Communications
Manager
Mm
When you are using site codes in mullisite environments with distributed call processing, call
processors must strip off the access and site code from the called number on outgoing calls. If
access and site codes are configured before the "." (dot) in the route pattern, you can easily strip
them offusing the discard digits instruction (DDI) on the route pattern orroute list. For
incoming calls, you need to use translation patterns to add the access code and appropriate site
tow
code that are used to get to ihe callersite.
Centralized Call-Processing
Deployments: Access and Site Codes
Cisco Unified
Communications
Manager
The example shows two sites with centralized call processing. Directory numbers in the main
site("headquarters." or "HO" in the figure) and the remote site(-'branch." or "BR'" in the
figure) partially overlap. Again, access and site codes are used losolve the problem of
overlapping directory numbers.
However, in this case, partitions and CSSs need lobedeployed ina way that phones at the
remote site do notsee director) numbers of main-site phones, and vice versa. Then a translation
pattern is added per site.
The translation pattern of each site includes the access and site code of the respective site.
Phones ateach site have a CSS assigned, which provides access tothe directory numbers of the
local site and the translation pattern for the other site orsites, fhetranslation patterns are
configured witha transformation mask that strips off the accesscode and site code. Further,
each translation pattern must have a CSS. which provides access to only those director)
numbers that are located at the target site ofthe respective translation pattern, fhis way., all
phones can dial local directory numbers and site-code translation patterns for accessing other
sites. After a user dials an intersite number(composed of the accesscode,site code, and
director)' number), the corresponding translation pattern is matched. The translation patlern
strips the site code and access code sothat only the directory number remains. This directory
number is matched again in the call-routing tableusing a CSS that hasaccess only to the
directory numbers of the site, which was identified bv the site code.
1-114 Implementing Cisco Unified Communications Manager. Part 2 (CIPT2) v8 0 12010 Cisco Systems. Inc.
Implementing PSTN Access in Cisco IOS
Gateways
This topic describes how to implement PSTN access in a multisiteenvironment.
When implementing PSTN access, the following digit manipulation has tobe performed b:fore
the call is sent outto thePSTN. Digit manipulation has to bedone in Cisco Unified
Communications Manager when anMGCP gateway isbeing used, and it can beperformed
either in Cisco Unified Communications Manager orat an H.323 gateway.
• Outgoing calls to the PSTN:
— Calling number transformation: Ifno direct inward dialing (DID) range isused at
the PSTN, transform all directory numbers to the same, single PSTN number in the
calling number. If DID is used, extend the directory numbers to a complete PSTN
number.
408 555-
DID XXXX 714 555-2222
PSTN
Incoming Cisco Unifed Communications Call from 9.1- Call from 714
Manager or gateway adds access code and 714 555-2222 555-2222 to 408
longdistance 1 to calling number. Cisco to 1001 555-1001
Unified Communications Manager or gateway
stnps PSTN prefixfrom called number.
As shown in the example, internal numbers have to be represented as valid PSTN numbers, and
PS'fN numbers should be shown with the accesscode 9 internally.
Note Adding the access code (and changing 10-digit PSTN numbers to 11-digit PSTN numbers,
including the long-distance "1" digit) tothe calling number ofincoming calls isnot required.
Adding it. however allows users to call back the number from call lists (such as received
calls or missed calls) without having toedit thenumber by adding therequired accesscode
ISDN TON
The TON isused tospecify in which format a number (such as calling number orcalled
number) isrepresented. Tohave a unique, standardized way to represent PSTN numbers in
Cisco UnifiedCommunications Manager, the numbers have to be transformed based on the
TON.
For example, ifthecalling number of an incoming PSTN call is received with a TON
subscriber, the PSTN access code can be prefixed so that the user can place a callback without
editing thenumber. Ifthe calling number is in national format, then the PSTN access code and
the national access code areprefixed. Ifa calling number is received with an international
TON. the PSTN access code and the international access code are prefixed.
In countries with fixed-length numbering plans, transforming the numbers is not required,
because users can identify the type ofcalling number that isbased on the length. In this case,
users can manually prefix the necessary access codes. In countries with variable-length
numbering plans, however, it can be impossible to identify whether thecallwas received from
the local area code, from another area code ofthe same country, orfrom another country by just
looking atthe number itself. In such cases, the calling numbers ofincoming PSTN calls have to
be transformed based on the TON.
^^1^4132673333
Incoming Calls
with Different
TONS
1001-1099
In the example, the main site gateway receives three separate calls, and callbacks should be
possible without requiring the user to edit the number. The first call is received from the local
area with a subscriber "ION and a seven-digit number. This number needs only to beprefixed
with access code9. Thesecond call,which is received with national TON and 10digits, is
modified by the addition ofaccess code 9and the long distance I,all ofwhich are required for
placing calls back to the source ofthe call. The third call is received from another country
(Germany, in this case) with an international TON. For this calk the access codes 9and 011
ha\c to be added to the received number, which begins with the country code.
1-118 Implementing Cisco Unrfied Communications Manager, Part 2(CIPT2) v8.0 >2010 Cisco Systems, Inc.
Implementing Selective PSTN Breakout
This topic describes how to implement selective PSTN breakout in a multisite environment.
"ITiere aretwo ways to select the local gateway forPSTN calls. One way is to configure a site-
specific set of route pattern, partition, CSS, route list, androutegroup. If you apply a site-
specific CSS at theend. a site-specific routegroup is used. This implementation model was the
only oneavailable before Cisco Unilied Communications Manager version 7.
With Cisco Unified Communications Manager version 7, thelocal route group feature was
introduced. With local route groups, all sites that share the same PSTN dial rules can use one
and the same route pattern (orsetof route patterns). Theroute pattern (orset of route patterns)
isput into a systemwide route list, and this route list includes the local route group. Atthe
device pool of thecalling device, oneof theconfigured route groups is configured to bethe
Standard Local Route Group for thiscaller. Inthismodel, the routegroupthat is used is
determined bythedevice pool of thecalling device and notby its CSS. Thelocal route group
feature simplifies dial plans because iteliminates the need for duplicate CSS, partitions, route
patterns, and route lists. Since local route groups have been introduced, they arethepreferred
method for local gateway selection.
IP phones located in the main site use the main-site gateway for
PSTN access. Remote phones use their local gateway:
• Single 9.@ route pattern.
• Route list configured to use local route group
• Phone device pool configured with route group to be used by phone.
- Mamsite device pool and remote site device pool refer to differentroute groups.
PSTN
Main Site
470 555-
1234 Remote Site
Cisco Unified
Communications
Manager
From a dial plan perspecti\ c.you create one 9Ar route pattern (assuming that the North
American Numbering Plan [NAN PI is used). This route pattern is in a partition that is part of a
global CSS that is used by all phones, fhe route pattern refers lo asystemwide route list that is
configured to use the local route group. At the site-specific device pools, the standard local
route group isset to the route group thai includes the site-specific gateway.
In the example, there would be a device pool for the main site and a device pool for the remote-
site. There would bea main site route group, including themain site gateway, and a remote site
route group, including the remote site gateway. IP phones at the main site and at the remote site
can now be configured with the same CSS. They all will match the same route pattern and
hence will use the same roule list. Based on the local route group feature, however, they will
alwavs use their local PSTN gateway for PSTN breakout.
Note The local route group is configured with NANP PreDot digit stripping, by default. If the H.323
gateway expects calls that arereceived from Cisco Unified Communications Manager and
that would be routed to the PSTN to includethe PSTN prefix 9, appropriate digit
manipulation has to beconfigured in Cisco Unified Communications Manager. In this case,
the best solutionwould be to configure the called-partytransformation patterns and apply
gateway-specific called-party transformation CSS atthe gateways.
PSTN Backup
If the IP WAN (ICT) fails, calls are rerouted over the PSTN:
• 8 (site code).XXXX route pattern per site.
• Route pattern points to route lists (first option: ICT; second option:
Local Route Group).
Remote Site
Main Site
Cisco Unified
Communications
Manager
1001-1099 1001-1099
The figure shows a multisite deployment with two sites. Each site has its own Cisco Unified
Communications Manager cluster. Intersite calls should use the intercluster trunk (ICT) over
the IP WAN. However, what if the IP WAN is down? Since both sites have access to the
PSTN, the PS'fN should be used as a backup for intersite calls.
To ensure that phones at different sites always use their local gateway for PSTN backup, a
route list is configured that includes the ICT as the first option and the local route group as the
second option. This way. there is no need to have multiple, site-specific route lists with a
difTerent. site-specific route group as second entry.
When you areusing PSTN backup for on-net calls,you must address internal versus external
dialing. While on-netcalls usual!) use site codes and directory numbers, calls that arc sent
through the PS'fN haveto use PSI'N fonnal. Digit manipulation requirements vary depending
on the path that i> taken for the call:
• Digit manipulation requirements when you use the ICT{first choice in route listand route
group):
— At the calling site: The accessand site codes are removed from called number.
— At the receiving site: The access and site codes are added to the calling number.
• Digit manipulation requirements when you use the PS'fN (secondary choice in roule list
and route group):
— At the calling site: The internal called number, which comprises an accesscode,
site code, and director} number, is transformed to the PSI'N numberof the called
phone. The calling number is transformed to the PSTN number of the calling phone.
Note IfDID is not supported, the PSTN number of the site, rather than the PSTN number of the IP
phone, is used in called number and calling number.
1-122 Implementing Cisco Unified Communicalions Manager, Part 2 (CIPT2) «8.0 '2010 Cisco Systems, Inc
At the receiving site: The PSTN callingnumberis recognized as a PSTN numberof
an on-net connected site and transformed to the internal number: access and site
code, followed by the directory numberof the callingphone (if DID is used at the
calling site)or of the attendant of the calling site (if DID is not used at the calling
site), The callednumberis transformed to an internal directory numberand routed to
the IP phone (if DID is used at the receivingsite) or to an attendant(if DID is not
used at the receiving site).
Implementing TEHO
Local Path
When you implement 11:HO. PSTN breakout occurs at the gateway that is closest to the dialed
PS 1N destination. Basically, this action occurs because you create a route pattern for each
destination area that can be reached at dilTerent costs, 'fhese route patterns refer to route lists
that include a route group for the TF.HO gateway first and the local route group as the second
entry so that the local gateway can be used as a backupwhen the IP WAN cannot be used.
Note The use of TEHO might not be permitted in your country or by your provider. There can also
be issues with emergency calls Therefore, ensure that your planned deployment complies
with legal requirements
1-124 Implementing Cisco Unifed Communications Manager. Part 2 (CIPT2) v8 0 © 2010 Cisco Systems, Inc.
Considerations for Using Remote PSTN Gateways
When using backup TEHO. you have to consider several potential issues.
The first thing to consider when you are using TEi 10 is what number you want to use for the
calling number of the outgoing call. Basically, there are two options for configuring the calling
number for the outgoing call:
• Use the PSTN number of the originating site at the TEHO gateway: When using the
PSTN number of the originating device for the caller ID of a TEHO call, the called party is
not aware that TEHO has been used. Standard numbering is maintained tor all PSTN calls,
regardless of the egress gateway; callbacks to the calling number are possible. Ilowcver,
sending calls lo the PSTN with PSTN caller IDs of other sites may not be permitted, or the
receiving PSTN provider may remove caller IDs from the signaling messages.
Caution Sending calls out of a gateway with the calling number of another site might not be permitted
in your country or by your provider. There can also be issues with emergency calls.
Therefore, ensure that your planned deployment complies with legal requirements.
• Replace the PSTN number of the originating site by the PSTN number of the TEHO
site: Whenusing the callingnumberof the backup gateway, called partiesmay get
confused about the number that should be used when calling back. For instance, they may
updatetheir address books with the differentnumberand inadvertently end up sendingcalls
to the TEHO site every lime ihey call. Further, DID ranges would have to include remote
phones or IVR scripts (automated attendants) to be able to route calls to phones located in
any site, regardless of where the PSTN call was received.
Caution Using a remote gateway for PSTN access mightnot be permittedin your country or by your
provider. There can also be issues with emergency calls. Therefore, ensure that your
planned deployment complies with legal requirements.
Note You must also consider Call Admission Control (CAC) when implementing TEHO. When the
primary (TEHO) path is not admitted, the local gateway should be used instead. More
information about CAC is provided in a separate module of this course.
1-126 Implementing Cisco Unified Communications Manager. Pari 2 (CIPT2) v8.0 ©2010 Cisco Systems, Inc.
TEHO Example Without Local Route Groups
This subtopic illustrates the dial plan requirements when you are implementing TEHO without
local route groups.
LI =j_
In the example, there are five sites in a centralized call-processing deployment. Each site uses
identical call-routing policies and numbering plans, but the site-specific details of those policies
prevent customers from provisioning a single set of route pattern and route list that works for
all sites. This principle applies when no local route groups are used (as it was the case before
Cisco Unified Communications Manager Version 7).
Although the primary path for a given TEHO PSTN destination is always the same (the
appropriate TEHO gateway), the backup path is different for each site (the local gateway of the
site where the call has been placed). Without a backup path, TEI10 would require only one
route pattern per TEHO destination number and would refer to only the corresponding TEHO
gateway from its route list and route group. However, as the IP WAN is used for TEHO calls, it
is not recommended that you configure a single path only. Therefore, TEHO configurations
easily end up in huge dial plans: Each site requires a different route pattern and route list for
each of the other sites. In addition, each site has one generic route pattern for non-TEHO PSTN
destinations (using the local gateway).
Note Some route patterns in the figure include the character"." multiple times (for example,
9.1 703.XXX.XXXX). In this case, theV character is used to illustrate the different
components of the number patterns in order to make it easier to interpret the patterns. In
reality, the"." in route patterns is used only once when being referenced by a corresponding
DDI, for example the PreDot DDI.
• For each TEHO route pattern, the first entry in the route list is the
TEHO location: the second entry is the local gateway.
• For non-TEHO destinations, there is only one entry (local gateway)
in the route list
* Forn sites, n ' n route patterns and route lists are required.
\>B'fW,
F« IB BI4. 1
In the example, the configuration for one site (Boulder) is illustrated. There is a TEHO route
pattern for area code 703 (Hemdon) that refers to the route list RE-Bldr-I Irdn. This route lists
uses the Hemdon gateway first and the (local) Boulder gateway as a backup. There is also a
route pattern for area code 972 (Richardson), again using a dedicated route list for calls from
Boulder to Richardson (with the Richardson gateway preferred over the local Boulder
gateway). 'fhere are tuo more such constructs for the other two sites. Finally, there is a generic
PSI'N route pattern (9/d) for all other PSTN (that is. non-TEHO) calls. The generic PSTN
route pattern refers to a route list that contains only the local gateway. All five route patterns
are in the Boulder partition (P-Bldr) so that they can be accessed only by Boulder phones
(using the Boulder CSS "CSS-Bldr").
In summan. for each TEHO destination there is a route pattern per originating site that refers to
a dedicated route list utilizing the appropriate TEHO gateway before the local gateway. For n
sites, there are n * (n - \) of these patterns. In addition, each site has a generic route pattern
referring lo a dedicated route list containing the local gateway only. This generic route pattern
increases the total number of route patterns and route lists to n * n. In large TEHO
deployments, this approach does not scale.
Note Some route patterns in the figure include the character"" multiple times (for example,
9.1.703 XXX XXXX) In this case, the "." character is used to illustrate the different
components of the number patterns in order to make it easier to interpret the patterns In
reality, the "." m route patterns is used only once when being referenced by a corresponding
DDI, for example the PreDot DDI
1-128 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) wB.O ) 2010 Cisco Systems. Inc.
TEHO Example with Local Route Groups
When implementing TEHO with local route groups, youcanreduce thenumber of route
patterns and route lists from n * n lo n + I.
M.7W.XXXJM* *|P«!«-»
t.i.lTs.sxjjaice lln^am
***!!*«(. I
This reduction is possible because, for each TEHO destination, one route pattern is sufficient.
The route pattern refers to a destination-specific route list, which lists the route group
containing the TEHO gateway first, followed by the entry "Default Local Route Group."
Because the backup path is now determined by the device pool of the calling device instead of
being explicitly listed in the route list, the route list has a generic format and can be used by all
sites.
For every TEHO destination, one route pattern and one route list is required. In addition, for
non-TFHO destinations, again, a single route pattern and route list can be utilized by all sites.
This route pattern (9.@) refers to a route list, which includes the "Default Local Route Group"
entry.
Note Some route patterns in the figure include the character V multiple times (for example,
9.1.703.XXX.XXXX). In this case, the "." character is used to illustrate the different
components of the number patterns in order to make it easier to interpret the patterns. In
reality, the "." in route patterns is used only once when being referenced by a corresponding
DDI, for example the PreDot DDI.
In the example (five sites), using local roule groups simpliiies the dial plan that is described
here:
• The number of route patterns and route lists for TEHO destinations is reduced from
n * In - 1) to n. In the example, the reduction is from 20 to 5.
• fhe number of route patterns and route lists for non-TEHO destinations is reduced from
n to 1 (5 to 1 in this example).
• Thus, the total number of route patterns and route lists is reduced from n * n to n + 1
(25 to 6).
The number of gateways. route groups, and de\ ice pools remains the same: ti.
1-130 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8 0 © 2010 Cisco Systems, Inc
Implementing Globalized Call Routing
This topic describes how globalized call routing is implemented and how it simplifies
international multisite dial plans.
With globalized call routing, all calls that involve extemal parties are based on one format. All
numbers are normalized as follows:
If sources of calls (users at phones, incoming PSTN calls at gateways, calls received through
trunks, and so on) do not use the normalized format, the localized call ingress must be
normalized before being routed, fhis requirement applies to all received calls(coming from
gateways and trunks, as well as from phones), and it appliesto both the calling-and called-
party numbers.
Note Except for the internal calls that were mentioned (where the destination is a directory
number and, inthe case of an internal source, the source is a directorynumber), all
numbers are normalized to the E.164 global format. Therefore, call routing that is based on
the normalized numbers is referred to as globalized call routing.
• Calling-party numbers for calls that are routed from gateways or trunks to phones:
This situation applies lo the phone user who does not want to sec caller IDs in a global
format. For example, ifa user at a U.S. phone wants to seethe numbers of PSTN callers
who arc in the same area code, that user may want to see each number as a seven-digit
number and not in the+1 XXXXXXXXXX fonnat.
Localized call egress is not needed for the called-party number of calls that arc routed to
phones, because internal directory numbers are the standard (normalized) fonnat for internal
destinations (regardless of the source of the call), fhese numbers might have been dialed
differenth initialK. however: in that case, this localized call ingress was nomialized before call
routing.
Localized call egress is also not required for the calling-party numberof internal calls (internal
to internal) because, again, the standard for the calling-party number of such calls is to use
internal director, numbers.
Globalized call routing simplifies international dial plans because the corecall-routing decision
is always based on the same fonnat. regardless of how the number was initially dialed and
regardless of how the number looks at the egress device.
1-132 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) vB.O ©2010 Cisco Systems, Inc
Globalized Call Routing: Number Formats
This subtopicdescribes the numberformats that are used by globalized call routingand
explains some commonly used expressions.
To External
Calling. Directory Number
Called According lo Loca
From Extemal:
Dal Rules (PSTN Access
Codes. International
Calling: Localized E.164
Called: Directory Number
Access Codes, Etc ) or
E 164 (Click to Dial, Call
Lists, Etc
From Internal: ^^Sg
Calling Directory Number
Called: Directory Number
To Interna
Calling Directory
Called. Directory
The table explains expressions that are commonly used to describe globalized call routing. The
table refers to the figure.
Term Description
Number The process of changing from normalized format (in this case, global
localization format) to local format. Usually, the local format is the shortest
possible format that does not conceal relevant information. An
example of local format is 555-1234 instead of +1 408 555-1234, or
972 333-4444 instead of +1 972 333-4444 (assuming that the device
where localization occurs is located in +1408 area).
Incoming PSTN Call from PSTN to internal phone. Like all calls, such a call consists
call of two call legs (incoming and outgoing). See also "Call ingress" and
"Call egress." On an incoming PSTN call, the incoming call leg (call
ingress) is PSTN gateway to Cisco Unified Communications
Manager; the outgoing call leg (call egress) is Cisco Unified
Communications Manager to internal phone.
Outgoing PSTN Call from internal phone to PSTN Like all calls, such a call consists
call of two call legs (incoming and outgoing). See also "Call ingress" and
"Call egress." On an outgoing PSTN call, the incoming call leg (call
ingress) is internal phone to Cisco Unified Communications Manager;
the outgoing call leg (call egress) is Cisco Unified Communications
Manager to PSTN gateway.
On the left side of the figure, call ingress is illustrated by two types of call sources:
• External callers: Their calls are received by Cisco Unified Communications Manager
through a gateway or trunk. In the case of a PSTN gateway, calling- and called-party
numbers are usually provided in localized E.164 format.
• Internal callers: Their calls are received from internal phones, in the case of calls to
internal destinations (for example, phone to phone), calling- and called-party numbers are
typically provided as internal directory numbers. In the case of calls to external destinations
(for example, phone to PS'fN). the calling number is the directory number (at call ingress
time) and the called number depends on the local dial rules for PS 1N access. These dial
rules can differ significantly for each location.
The center of the figure illustrates the standards that are detlned for normalized call routing. As
mentioned earlier, because most calls use global E.164 format, this type of call routing is also
referred to as globalized call routing. Ilere are the defined standards:
• External to internal:
1-134 ImplementingCisco Unified Communications Manager, Part 2 (CIPT2)v8.0 © 2010 Cisco Systems. Inc
At the right side of the figure, call egress is illustrated by two types of call targets:
• Gateways: When sending calls to the PSTN, localized E.164 format is used for both the
calling- and called-party numbers. The format of these numbers (especially of the called-
party number) can significantly differ based on the location of the gateway. For example,
the international access code in the United States is 011, and in most European countries, it
is 00.
• Phones: When a call from an internal phone is sent to another internal phone, the call
should be received at the phone with both the calling and called number using internal
directory numbers. Because this format is the same format that is used by globalized call
routing, there is no need for localized call egress in this case. When a call from an external
caller is sent to an internal phone,most users (especially users in the United States)prefer
to see the calling number in localized format (for example, national and local calls should
be displayedwith 10digits). The called numberis the directorynumberand usually is not
displayed on the phone.
It is evident from the figure that there are several situations where the numbers that are
provided at call ingress do not conform to the normalized format to be used forcall routing.
These situations applyalso to call egress, wherethe normalized format is not always used when
the call is delivered. Therefore, localized call ingresshas to be normalized (that is. globalized)
and globalized fonnat has to be localizedat call egress.
Cisco Unified
Communicalions
Called Number
Manager
Globalized
Call Routing
Ilere are the requirements for normalizing localized call ingress on gateways:
• Changing the calling number from localized E.164 fonnat to global E.164 fonnat
• Changing the called number from localized E.164 fonnat to directory numbers for calls to
internal destinations
• Changing the called number from localized E.164 formal to global E.164 fonnat for calls to
external destinations (if applicable)
As shovsn in the figure, the calling number canbe normalized by incoming calling-party
settings. Thev are configured at the gateway oralthe device pool, orthey can be configured as
CiscoUnified Communications Manager serviceparameters. The figure provides an example
for a gateway in San Jose:
• Prefix for incoming called-party numbers with number typesubscriber: +1408
• Prefix for incoming called-party numbers with number type national: +1
• Prefix for incoming called-party numbers with number type international: +
The called numbercan be normalized bv significant digits that are configured al the gatewav
(applicable onlv ifno calls lo other external destinations are permitted and a fixed-length
number plan is used), or bv translation patterns, orby incoming called-party settings (if
available at the ingress dev ice). In the example, the gateway is configured with four significant
digits.
1-136 Implementing Cisco Unified Communications Manager, Part2 (CIPT2) vS.O >2010 Cisco Systems, Inc
Normalization of Localized Call Ingress from Phones
The figure illustrates how localized call ingress on phones gets normalized.
Here are the requirements for normalizing localized call ingress on phones:
• For calls to external destinations: Changing the calling number from an internal director}
number to FT 64 fonnat. Changing the called number to K.164 format if any other format
was used (according to local dial rules).
• For calls to internal destinations: No normalization is required.
As shown in the example, you can normalize the calling-party number for calls to external
destinations by configuring an extemal phone number mask (in E.164 format) at the phone.
You can normalize the called-party number by using translation patterns where you would also
apply the extemal phone number mask to the calling-party number. In the figure, examples for
phones that are located in Hamburg, Germany, and San Jose, California, are given.
Called Number
Called Parly
Cisco Unified
Transformation CSS
Communications
(GW. DP)
Manager
Globalized
Calling Number
Call Routing
Calling Party
Trans form a lion CSS
(GW. OP)
The only requirement is to change the calling and called number from global F.164 format to
localized F., 164 fonnat.
You can change the format by configuring called- and calling-party transformation patterns,
puttingthem into partitions, and assigning the appropriate called-and calling-party
transformation CSS to gateways. You can configure called- and calling-party transformation
CSS at the de\ice (gateway or trunk) and al the device pool.
The tables that are presented in this section refer to the example that is provided by the figure.
Hie first table shows the configuration ofthe called-party transfoniiation patterns that are
applicable to the SanJosegateway (based on partition andcalled-party transformation CSS).
Note In this example, the San Jose gateway does not use number types. Therefore, 011 has to
be prefixed on international calls, and the 1 of national calls is conserved. Forlocal calls,
only the last seven digits are used.
1-138 Implementing Cisco Unified Communications Manager. Part 2 (CIPT2)v8 0 © 2010 Cisco Systems, Inc.
The next table shows how you would configure the called-party transformation patterns that are
applicable to a gateway in Hamburg, Germany (based on partition and called-party
transformation CSS).
Note In this example, the Hamburg gateway is using number types instead of international (00) or
national (0) access codes (in contrast to the San Jose gateway, which does not use number
types).
The next table shows how you would configure calling-party transformation patterns that are
applicable to the San Jose gateway (basedon partitionand calling-party transformation CSS).
Note In the example, subscriber, national, and international number types are used at the San
Jose gateway for the calling-party number. If no number types were used, due to the fixed-
length numbering plan, the number type could also be determined by its length (seven-digit
numbers when the source of the call is local, 10-digit numbers when the source of the call is
national, or more than 10 digits when the source of the call is international). In reality,
however, countries that use the NANP typically use 10-digit caller IDs for both national and
local callers
Having nonlocal calling-party numbers implies the use of TEHO or PSTN backup over the IP
WAN. This scenario is not permitted in some countries or by some PSTN providers. Some
providers verify that the calling-party number on PSTN calls that they receive matches the
locally configured PSTN number. If a different PSTN number is set for the caller ID, eit.ier
the call is rejected or the calling-party number is removed or replaced by the locally
assigned PSTN number
The final table shows how you would configure calling-party transformation patterns that are
applicable to a gateway in Hamburg, Germany (based on partition and calling-party
transformation CSS).
The only requirement is that you change the calling number from global li. 164 fonnat to
localized F.164 fonnat.
You can change the fonnat by configuring calling-party transformation patterns, putting them
into partitions, and assigning the appropriate calling-parly transformation CSS to IP phones. As
mentioned earlier in this lesson, you can configure calling-party translomialion CSS at the
phone and at the de\ ice pool.
The two tables that are presented in this subtopic arc in reference to the example that is
provided by the figure. The first table shows how you would contigurc the calling-party
transformation patterns that are applicable to a phone that is located in San Jose (based on
partition and calling-part} transformation CSS).
Note In this example, international calls are shown in standard normalized format (E 164 format
with + prefix) because there is no W calling-party transformation pattern. National calls are
shown with 10-digit caller IDs, and local calls are shown with 7-digit caller IDs.
1-140 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2)v8 0 © 2010 Cisco Systems, Inc
The next table shows how you would configure the calling-party transformation patterns that
arc applicable to a phone that is located in Hamburg, Germany (basedon partitionand calling-
party transformation CSS).
Note Because there is no \+! calling- party transformation pattern, international calls are
preserved in normalized format (E.164 with + prefix). As opposed to the San Jose example,
phones that are located in Hamburg do prefix the national access code (using 0, which is
equivalent to the long-distance 1 in the NANP). The reason is that, in Germany, variable-
length PSTN numbering plans are used and therefore national and local numbers cannot be
distinguished based on their length (like in the United States, with 7- and 10-digit numbers).
When the national access code 0 is prefixed to numbers that are used by national callers, a
user can identify national calls by their leading 0.
Note When users call back PSTN callers, the globalized number is used for the outgoing call.
Therefore, there is no need to edit the localized number from a call list and add PSTN
access codes and national or international access codes.
Note local emergency numbers (for example, 112 in the European Union,
999 in the United Kingdom. 000 in Australia).
Introduce a corporate emergency number (for example, 888) that can be
used at all sites (globalized emergency number).
In addition, however, localized emergency dialing should still be supported, so that a user can
dial eitherthe locally rele\ ant emergency numberor the corporate emergency number.
Here is how to implement such a solution;
• You introduce one or more corporate emergency numbers.
• In addition. \ou allow localized emergency dialing. It can he limited to local emergency
dialing rules persite (for example, an Austrian emergency number can be dialed only from
phones that are located in Austria), or you can globally enable all possible local emergency
numbers. Having all possible local emergency numbers thatare globally enabled would
allow a roaming user to use the emergency number that is local to thesite where the user is
located, or theemergency number thatthe userknows from the home location of the user
(forexample, a UK userdials 999 while roaming in Austria), or the corporate emergency
number.
• Ifa user dials a localized emergency number, that numberis first normalized (that is.
translated) to the corporate emergency number. A route pattern exists only forthis
corporate emergency number, and you configure the corresponditig route listto use the
local route group.
1-142 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 © 2010 Cisco Systems, Inc.
• At the gateway that is used to process the call, you localize the corporate emergency
number (the globalized emergency number) by using called-party number transfonnations
at the gateway. This localization ensures that, regardless of which emergency number was
dialed, the gateway that sends out the emergency call uses the correct number as expected
at this site.
Note In deployments with more complex emergency calls, like in the United States with E911,
such a solution is not applicable because there are other requirements for emergency calls.
In such a scenario, the emergency call is routed via a dedicated appliance (Cisco
Emergency Responder) that is reached via a computer telephony integration (CTI) route
point.
Called-Party
000 Route Transformation
Pattern
Cailed-Party / *
nn
Transformation -' :' Route Lisl
Mask BBS
Syslem
Roule • 33 Called-Party
Transformation
Group
Default UK Gateway 8B8 ^ 999
LRG
Any User in UK
In the example, a corporate emergency number of 888 has been established. In addition.
Australian. E.U.. and UK emergene; numbers are supported at all sites oflhe enterprise. The
appropriate numbers (000. 112. and 999) are translated (normalized) to the corporate (global)
emergency number 888. A route pattern888 exists, which refersto a roule lisl that has been
configured to use the local routegroup. You will considertwo sites in this example: one in the
European Union and one in the United Kingdom. Hach site has its own PSTN gateway (or
"GW" in the figure): phonesat each site are configured with a site-specific device pool. The
de\ ice pool of each site has its local routegroup that is set to a site-specilic route group.
You will examine fouremergenc} calls:
• A IK user dials 999 (IK emergency number):
The dialed UK emergency number999 is translated lo the corporate emergency number
888. After translation, the 888 route pattern is matched. The route list of the route pattern
refers to the local routegroup. Because the emergency call was placed from a UK phone,
the local routegroup in the devicepool of the phone refers to the UK gateway. At thai
gateway, a global transformation of the called number (from 888 to 999) is configured.
Therefore, the call exits the UK gateway with a destination numberof 999, which is the
appropriate emergency number to be used in the United Kingdom.
• Any user who is located in the United Kingdom dials888 (corporate emergency
number):
Because no local emergency number was dialed exceptthe corporate emergency number
888. no translation is required. The call immediately matches routepattern 888. The route
listof the route pattern refers to the local route group. Because theemergency call was
placed from a UK phone, the local route group in the device pool of the phone refers to the
UKgate\sa\. Atthat galewa\. a global transformation of the called number (from 888 to
999) isconfigured, "fherefore. the call exits the UK gateway with a destination number of
999. uhich is the appropriate emergency numberto be used in the United Kingdom.
1-144 ImplementingCisco Unified Communications Manager. Part 2 (CIPT2)v8.0 © 2010 Cisco Systems, Inc
• An E.l. user dials 112 (E.U. emergency number):
The dialed E.U. emergency number 112 is translated to the corporate emergency number
888. After translation, the 888 route pattern is matched. The route list of the route pattern
refers to the local route group. Because the emergency call was placed from an E.U. phone,
the local route group in the device pool of the phone refers to the E.U. gateway. At that
gateway, a global transformation of the called number (from 888 to 112) is configured.
Therefore, the call exits the E.U. gateway with a destination number of 112, which is the
appropriate emergency number to be used in the E.U.
• An Australian user, currently located at an E.U. site, dials 000 (Australian emergency
number):The dialed Australian emergency number 000 is translated to the corporate
emergency number 888. After translation, the 888 route pattern is matched. The route list
of the route pattern refers to the local route group. Because the emergency call was placed
from an E.U. phone, the local route group in the device pool of the phone refers to the E.U.
gateway. At that gateway, a global transformation of the called number (from 888 to 112)
is configured. Therefore, the call exits the E.U. gateway with a destination number of 112,
which is the emergency number in the European Union.
Note The Australian user can use an E.U. phone (with an E.U. extension), or use their own device
with device mobility enabled, or use an E.U. phone with their own extension (by using Cisco
Extension Mobility). In all three scenarios, the emergency call would work fine as described
earlier. The reason is that the device pool of the phone will be the E.U. device pool in all
three scenarios (with device mobility enabled, the home device pool would be replaced by
the roaming device pool), and hence the local route group is always the EU-GW.
The only problem would be if the Australian user were using their own device with device
mobility disabled. In this case, the local route group would refer to the Australian gateway,
and therefore the call would be sent through the Australian gateway instead of through the
local E.U. gateway. The localized egress number would be appropriate for an Australian
gateway (transformed to 000), so that the user would get connected to an Australian
emergency service.
•AAR
• SRST or CFUR
1-146 Implementing Cisco Unifed Communications Manager. Part 2 (CIPT2)v8.0 ) 2010 Cisco Systems, Inc.
Globalized Call Routing—TEHO Advantages
fhis subtopic reviews the advantages of using globalized call routing in an international dial
plan that uses TEHO.
As discussed earlier,when you are using local route groups, there is no need to have duplicated
TEHO route patterns for each originating site. Instead, the local PS'fN gateway is selectedby
the local route group feature when the TEHO path cannot be used.
Whencombining globalized call routing with local route groups,you do not have to care about
the variouspossible input formats for the TEHO call-routing decision. No matter how the user
dialed the number, it is changed to globalized format before it is routed. Because the called
number is then localized after call routing and path selection, you can localize the called- and
calling-party number differently at the primary gateway (TEHO gateway) and the backup
gateway (localgateway). However, the global transformations that you configure for each
egress gateway all refer to a single format—a globalized format regardless of how the user
dialed the destination. This globalized format that is combined with local route groups for local
backup gateway selection, makes implementing TEHO much simpler. Without globalized call
routing, youwould haveto perform localization at the egress gateway differently foreach
originating site.
At the call ingress side, there are three PSTN dial rules: E.U.. UK,and United StatesThe same
rules applv to the egress gateways: the E.U.. UK, and U.S. gateways all require dilTerent digit
manipulation when you are sending calls to the PSTN.
As long as users areallowed to roam between sites and TEHO with local backup is inplace,
users can dial each PSTN destination differently at each site. In addition, if the TEHO path is
notavailable, the local gateway (which again can be anyof the three) is used for backup. With
globalized call routing, vou do not have to consider all possible combinations ofingress and
egress, butyou consider call ingress and call egress independent of each other.
All that vou need to configure is translation patterns for each of the PSTN dial rules (E.U.. UK.
and United States), "fhen vou create TEI10 route patterns that refer to the TEHO gateway as the
first choice, and to the local gateway as the backup, using the local route group feature. At the
egress gatewavs. vou configure the called- and calling-party transfonnations, where you do not
match on all possible input formats again, buton a globalized format only.
1-148 Implementing Cisco Unified Communications Manager. Part 2 (CIPT2) v8.0 © 2010 Cisco Systems, Inc
Summary
fhis topic summarizes the keypoints that werediscussed in this lesson.
Summary
Summary (Cont.)
• Cisco Svstems. Inc. Cisco UnifiedCommunications System 8.x SRND, April 2010.
http:/.•''www.cisco.coni'en'US'VJoes'voice ip comm/cuem/srnd/Xx.'uc8x.html
• Cisco Systems, Inc. Cisco Unified ( ommunications Manager Administration Guide
Release'8.0(1/. Eebruary 2010.
http:"'www cisco.com.civI'S'docs'voice ip comm/cuem/admin/8 0 |/ccmdg/hccin-80l-
cm.htinl
• Cisco Svstems. Inc. Cisco IOS Voice Configuration Library (with Cisco IOS Release 15.0
updates). July 2007.
http:;\\v\vv.cisco.coni'cnT'S'doc.s/ios'!2 .Vvvf c'cisco ios voice configuration librarv £
lossarv/vcl.htm
1-150 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) vS.O ©2010 Cisco Systems. Inc
Module Summary
lliis topic summarizes the key points that were discussed in this module.
Module Summary
This module discussed the issues thatapply to Cisco Unified Communications Manager
multisite deployments and their possible solutions. Itdescribed thevarious connection options
for muitisite deployments and how they are implemented. Itthen described how to implement a
multisite dial planthat covers site-code dialing, public switched telephone network (PSTN)
backup, and tail-end hop-off (TEHO).
References
For additional information, refer to these resources:
• Cisco Systems. Inc. Cisco Unified Communications System 8.xSRND, April 2010.
http:.'7'vvvvw.cisco.com/en/US/docs/voice_ip_c()mm/cucm/snid/8x/uc8x.htnil
• CiscoSystems, Inc. Cisco Unified Communications Manager Administration Guide
Release 8.0(1). February 2010.
hitp:/.'vv-w-w.cisco.com/cn/US/docs/voice_ip_comm/cucm/adinin/8 0_l/ccmcfg/bccm-80l-
cm.html
• Cisco Systems. Inc. Cisco IOS Voice Configuration Library (with Cisco IOS Release 15.0
updates). July 2007.
hup; 7ww w.cisco.com/'en/US/does/ios/l2..3/vvf_c/cisco. ios_voice...configuration librarv_g
lossaryvcl.htm
Q2) Which of these statements does not apply to IP networks? (Source: Identifying Issues
in a Multisite Deployment)
A) IP packets can be delivered in incorrect order.
B) Buffering results in variable delays.
C) Tail drops result in constant delays.
D) Bandwidth is shared by multiple streams.
Q3) Which statement most accurately describes overhead for packetized voice? (Source:
Identifying Issues in a Multisite Deployment)
A) VoIP packets are large and sent at a high rate.
B) The Layer 3 overhead of a voice packet is not significant.
C) Voice packets have small payload size and are sent at high packet rates.
D) Packetized voice has the same overhead as circuit-based voice.
Q6) Which of these is a requirement for performing address translation for Cisco IP
tmm phones? (Source: Identifying Issues in a Multisite Deployment)
A) use DHCP instead of fixed IP addresses
B) exchange media streams with the outside world
C) use DNS insteadof hostnames in Cisco UnifiedCommunications Manager
D) exchange signaling information with the outside world
Q8) When implementing QoS. how is the quality of voice streams provided? (Source:
Identifying Multisite Deployment Solutions)
Q9) Which two statements are tme about bandwidth solutions in a multisite deployment?
(Choose two.) (Source: Identifying Multisite Deployment Solutions)
A) RTP-header compression compresses the RTP header to 2 bytes.
B) WAN bandwidth can be conserved by using low-bandwidth codecs within a
remote site.
C) WAN bandwidth can be conserved by deploying local media resources.
D) Voice compression is part of RTP-header compression.
E) Multicast MOH from branch router flash totally eliminates the need to send
MOH over the WAN.
QIO) Which tuo statements are tme about availability? (Choose two.) (Source: Identifying
Multisite Deployment Solutions)
A) CFNB is required to enable main-site phones to call remote-site phones during
SRST fallback.
B) SRS'f provides a fallback for Cisco IP phones.
C) MGCP fallback allows the gateway to use local dial peers when the call agent
is not reachable.
D) AAR is required to enable phones to reroute calls over the PS'fN when the IP
WAN is down.
F) MGCP fallback and SRS'f cannot be implemented at the same device.
Ql 1) Whichof the following are notdial-plan solutions for multisite Cisco Unilied
Communications Manager deployments? (Choose two.) (Source: Identifying Multisite
Deplovment Solutions)
A) access and site codes
B) TEHO
C) globalized call routing
D) shared lines
E) overlap signaling
QI2) Which Cisco IOS feature provides signalingand media proxy functionality in order to
eliminate the need for NAT? (Source: Identifying Mullisite Deployment Solutions)
A) Cisco Unilied Border Element in flow-through mode
B) Cisco PIX Firewall
C) Cisco Unitied IP-to-Proxy Gateway
D) Cisco Unilied Border Element in flow-around mode
1-154 implementing Cisco Unified Communications Manager. Pari 2 (CIPT2) v8.0 ©2010 Cisco Systems, Inc.
Q13) Whichofthe following is nota connection option for a multisiteCisco Unified
Communications Managerdeployment? (Source: Implementing Multisite Connections)
A) SIP trunk
B) SIP gateway
C) H.323 gateway
D) H.225 trunk
Q14) Which two commands are required to enable MGCP at a gateway when using the
configuration server feature? (Choose two.) (Source: Implementing Multisite
Connections)
A) mgcp
B) seep
C) ccm-manager config server ip-address
D) ccm-manager config
E) ccm-manager seep
Q15) Which parameter is set in the H.323 gateway configuration window in order to strip the
called party number to a certain number of digits? (Source: Implementing Multisite
Connections)
A) Called Party Digits Mask
B) Significant Digits
C) Calling Party Transformation Mask
D) Number-Length
F) Discard Digit Instruction
QI6) Which two tmnks are not configured with the IP address ofthe next signaling device in
the path? (Choose two.) (Source: Implementing Multisite Connections)
A) H.225 trunk
B) nongatekeeper-controlled intercluster trunk
C) SIP trunk
D) gatekeeper-controlled intercluster tmnk
E) MGCP trunk
Q17) Where do you configure SIP timers and features for an SIP trunk? (Source:
Implementing Multisite Connections)
A) SIP profile
B) SIP security profile
C) SIP tmnk security profile
D) common trunk profile
Q18) Which ofthe following needs to be specified in the gatekeeper configuration window
when adding a gatekeeper to Cisco Unified Communications Manager? (Source:
Implementing Multisite Connections)
A) H.323 ID of the gatekeeper
B) IP address ofthe gatekeeper
C) zone name
D) technology prefix
Q22) fhe PSTN egress gateway can be selected in which two of these ways? (Choose two.)
(Source: Implementing a Dial Plan for International Multisite Deployments)
A) by the partition ofthe calling device
B) based on the CSS ofthe gateway
C) bv the local route group feature
D) based on the matched roule pattern when route patterns exist once per site
F) by the standard local routegroup that is configured at the gateway device pool
Q23) Where can digit manipulation be performed when digit manipulation requirements vary
for the on- and off-net paths? (Source: Implemenlinga Dial Plan for International
Multisite Deployments)
A) per route group ofthe route list
B) route pattern
C) directorv number
D) translation pattern
Q24) When implementing TEHO for national calls and using the local PSTN gateway as a
backup, how many route patterns arerequired fora cluster with three siteslocated in
different area codes?(Source: Implementing a Dial Plan for International Multisite
Deplov ments)
A) 3. when not using the local route group feature
B) 6. when using the local route group feature
C) 9. when not using the local route group feature
D) 4. when using the local route group feature
1-156 Implementing Cisco Unitied Communications Manager. Part2 (CIPT2) v8 0 © 2010 Cisco Systems, Inc.
Q25) Which of these is used to globalize the callingparty number of inbound PSTN calls?
(Source: Implementing a Dial Plan for International Multisite Deployments)
A) globalization type
B) called number
C) inbound gateway identifier
D) number type
Q26) The implementation of globalized call routing does not simplify the deployment of
which two of these features? (Choose two.) {Source: Implementing a Dial Plan for
International Multisite Deployments)
A) TEHO
B) Device Mobility
C) AAR
D) MOH
E) Cisco Extension Mobility
F) SRST
G) local conference bridges
02) C
Q3i C
O-ti D
05) A.C
061 B
Q7) D
OS) Voice packets are given absolute pnority over other traffic
Q9) C. V:
QIO) B.C
qui I).)-.
012) A
013) B
014) C. D
Q15> B
016) A. D
017) A
Q18) B
019) C
Q20) A
0^1) D
022) c. n
02?) A
024) D
025) D
Q26) D.G
1-158 Implementing Cisco Unified Communications Manager. Part 2 (CIPT2) v8 0 ©2010 Cisco Systems, Inc
Module 2
Centralized Call-Processing
Redundancy Implementation
Overview
The capability to use centralized call-processing devices that are located at remote sites
depends on the availability of Cisco Unified Communications Manager at the main site. To
provide these devices with a backup, you can use Media Gateway Control Protocol (MGCP)
fallback and Cisco Unified Survivable Remote Site Telephony (SRST).
This module describes the mechanisms for providing call survivability and device failover in
remote sites. It describes how to configure Cisco IOS routers as Cisco Unified SRST gateways
and how to use Cisco Unified Communications Manager Express in Cisco Unified SRST mode.
Module Objectives
Upon completing this module, you will be able to implement call-processing resiliency in
remote sites by using Cisco Unified SRST, MGCP fallback, and Cisco Unified
Communications Manager Express in Cisco Unified SRST mode. This ability includes being
able to meet these objectives:
• Describe the mechanisms for providing call survivability and device failover in remote
sites, including the functions, operation, and limitations of each mechanism
• Configure Cisco Unified SRST to provide call survivability for IP phones, and MGCP
fallback for gateway survivability
• Configure Cisco Unified Communications Manager Express to provide telephony services
to IP phones if the connection to the centralized call agent is lost
2-2 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v80 ©2010Cisco Systems, Inc.
Lesson 1
Objectives
Upon completing this lesson, youwill be able to describe the mechanisms forproviding call
survivability and device failover inremote sites, including thefunctions, operation, and
limitations of each mechanism. This ability includes being able to meet these objectives:
• Describe remotesite redundancy optionsand comparetheir characteristics
• Describe how Cisco Unified SRST works
• Describe Cisco Unified SRST versions, their protocol support, their features, and the
required Cisco IOS Software releases
• Describe dial plan requirements for MGCP fallback and Cisco UnifiedSRST
Remote Site Redundancy Overview
This topic describes the various technologies that are used toprovide remote site redundancy
for small and medium remote sites in a Cisco Unified Communications Manager environment.
Cisco Unified SRS'f and MGCP gatewav fallback are the key components to the delivery of
fail-safe communication services.
Cisco Unifed
Communications
Manager
Cisco Unified Communications Manager supports Cisco Unitied IP phones at remote sites that
are attached to Cisco multiservice routers across the WAN. Before Cisco Unified SRST was
available, when the WAN connection between a router and the Cisco Unified Communications
Manager failed or when connectivity with Cisco Unified Communications Manager was lost.
Cisco Unified IP phones on the network became unusable for the duration ofthe failure.
Cisco Unified SRST overcomes this problem and ensures that Cisco Unified IP phones offer
continuous (although minimal) service by providing call-processing support for Cisco Unified
IP phones directly from the Cisco Unified SRS'f router. The system automatically detects a
failure and uses Simple Network-Enabled Auto Provision (SNAP) technology to autoconfigure
the branch office router to provide call processing for Cisco Unified IP phones that are
registered with the router. When the WAN link or connection to the primary Cisco Unitied
Communications Manager subscriber is restored, call processing reverts lo the primary Cisco
Unified Communications Manager.
MGCP gatewav fallback is a mechanism that allows a Cisco IOS router to continue to provide
voice gateway functions even when the MGCP call agent is not in control ofthe media
gateway, fhese voice gatewav functions are implemented through a fallback mechanism that
activates the so-called default technology application. The gateway then works in the same way
as a standalone H.323 or Session Initiation Protocol (SIP) gateway.
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 ©2010 Cisco Systems. Inc
Remote Site Redundancy Technologies
'fhe table lists the capabilities ofdifferent remote site redundancy technologies.
To use Cisco UnifiedSRSTas your fallback modeon an MGCP gateway, you must configure
Cisco Unified SRST and MGCP fallback on the same gateway. MGCP and Cisco Unified
SRST have had the capability to be configured on the same gateway since Cisco IOS Software
Release 12.2(11)T.
Cisco Unified SRSTalso provides a basic set of featuresto SIP-based IP phones. This set of
Cisco Unified SRST basic features is also known as Cisco Unified SIP SRST. Cisco Unified
SIP SRST has to be enabledand configured separately on Cisco IOSrouters. Cisco Unified
SRST versions 3.3 and earlier provide a SIP Redirect Server function; in subsequent versions,
this function acts as a back-to-back user agent (B2BUA).
Cisco Unilied Communications Manager Express in Cisco Unified SRST mode provides more
features to a smaller maximum number of IP phones by falling back to Cisco Unified
Communications Manager Express mode. The main feature enhancements include presence,
Cisco Extension Mobility, and support of local voice-mail integrations.
VoIP call preservation sustains connectivity for topologies in which signaling is managed by an
entity (such as Cisco Unified Communications Manager) that is different from the other
endpoint and that brokers signaling between the two connected parties.
Call preservation is useful when a gateway and the other endpoint (typically a Cisco Unifi ;d IP
phone) are eolocated at the same site and the call agent is remote. In such a scenario, the call
agent, the gateway with the remote endpoint, will more likely experience connectivity failures.
Cisco Unified Communications Manager Express version 8.0 supports a maximum of 450 IP
phones (Cisco IOS 3945E router) while Cisco Unified SRST version 8.0 supports up to 1500 IP
phones on the same platform. Refer to "Cisco Unified Communications Manager Express 8.0"
(http://v\^^v.eisco.coni/en.TJS/prod/collatcral/voiccsvv/ps6788/vcallcon/ps4625/data_shcct_c78-
567246.html) for more details about the supported number of phones.
Although MGCP gatewav fallback is most often used together with Cisco Unified SRST to
provide gateway functions to IP phones in Cisco Unified SRS'f mode, it can also be used as a
standalone feature. One example is that for a fax application server that uses a PRI ISDN
interface that is controlled by MGCP, connectivity to the PSTN can be preserved by MGCP
gateway fallback. Another example of an MGCP-fallback standalone configuration is a
mechanism that allows analog interfaces that arc controlled by Skinny Client Control Protocol
(SCCP) to stay in service even when the WAN connection to the Cisco linified
Communications Manager is down.
MGCP gatewav fallback preserves active calls from remote site IP phones lo the PSTN when
analog or channel associated signaling (CAS) protocols are used, for ISDN protocols, call
preservation is impossible, because Layer 3 ofthe ISDN stack is disconnected from the MGCP
call agent and is restarted on the local Cisco IOS gateway. Consequently, for active ISDN calls,
all call-state infonnation is lost in cases of switchover to fallback operation.
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 ©2010 Cisco Systems. Inc
When to Use Cisco Unified SRST
Cisco Unified SRST provides Cisco Unified Communicalions Manager with fallback support
for Cisco Unified IP phones that are attached to a Cisco router ona local network.
Cisco Unified SRST enables routers to provide basic call-processing support for Cisco Unified
IP phones when they loseconnection to remote primary, secondary, and tertiary Cisco Unified
Communications Manager installations or when the WAN connection is down.
Cisco UnifiedSRSf also supportssecurity features. If IP phones are configured with security
mode authenticated or encrypted in Cisco Unified Communications Manager and secure Cisco
Unified SRST is deployed, securityfeatures of Cisco IP phones are preservedduring fallback.
Cisco Unified SRSTcan support SIP phoneswith the standard RFC 3261 feature locallyand
across SIP WAN networks. With Cisco Unified SIP SRST, SIP phones can place calls across
SIP networks in the same way that SCCP phones do.
Cisco Unified SIP SRST supports the following call combinations: SIP phone to SIP phone,
SIP phone to PSTN or router voice port, SIP phone to SCCP phone, and SIP phone to WAN
VoIP using SIP.
SIP proxy, registrar, and B2BUA servers are key components of a SIP VoIP network. These
servers are usually located in the core of a VoIP network. If SIP phones that are located at
remote sites at the edge ofthe VoIP network lose connectivity to the network core (because of a
WAN outage), they may be unable lo make or receive calls. Cisco Unified SIP SRST
functionality on a SIP PSTN gateway provides service reliability for SIP-based IP phones in the
event of a WAN outage. Cisco Unified SIP SRST enables the SIP IP phones to continue to
make and receive calls to and from the PSTN and also to make and receive calls to and from
other SIP IP phones.
When the IP WAN is up. the SIP phone registers with the SIP proxy server and establishes a
connection to the B2BUA SIP registrar (B2BUA router). But any calls from the SIP phone go
to the SIP proxy server through the WAN and out to the PSTN.
Note The B2BUA acts as a user agent to both ends ofa SIP call. The B2BUA is responsible for
managing all SIP signaling between both ends of the call, from call establishment to
termination Each call is tracked from beginning to end, allowing the operators of the B2BUA
to offervalue-added features to the call. To SIP clients, the B2BUA acts as a user agent
server on one side and as a user agent client on the other (back-to-back) side. The baste
implementation of a B2BUA is defined in RFC 3261.
Cisco Unified SRSTdoes not supportenhanced features such as presence or Cisco Extension
Mobility. Message Waiting Indicator (MWI) is also not supported in fallback mode.
2-8 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 © 2010 Cisco Systems, Inc
When to Use Cisco Unified Communications Manager Express
in SRST Mode
Cisco Unified Communications Manager Express in SRST mode enables routers to provide
call-processing support to Cisco Unified IP phones by Cisco Unified Communications Manager
Express ifthe phones lose connection toremote primary, secondary, and tertiary Cisco Unified
Communications Manager installations or if the WAN connection is down.
Main Site
Cisco Unified
Communications
Manager
-W.N
Cisco Unified Communications Manager supports Cisco Unified IP phones al remote sites that
are attached to Cisco multiserv ice routers across the WAN. fhe remote site IP phones register
with Cisco Unitied Communications Manager. Keepalive messages are exchanged between IP
phones and the central Cisco Unified Communications Manager across the WAN. Cisco
Unified Communications Manager at the main site manages the call processing for the branch
IP phones.
2-10 Implementing Cisco Unified Communications Manager, Part 2 (C1PT2) v8.0 ©2010 Cisco Systems, Inc
Cisco Unified SRST Function: Switchover Signaling
When Cisco Unified IP phones lose contact with Cisco Unified Communications Manager, they
register with the local Cisco Unified SRST router tosustain the call-processing capability that
is necessary to place and receive calls.
Cisco Unified SRST configuration provides the Cisco Unified IP phones with the alternative
call control destination ofthe Cisco Unified SRST gateway.
When the WAN link fails, the Cisco Unified IP phones lose contact with the central Cisco
Unified Communications Manager but then register with the local Cisco Unified SRST
gateway.
The Cisco Unified SRST gateway detects newly registered IP phones, queries these IP phones
for their configuration, and then autoconfigures itself. The Cisco Unified SRST gateway uses
SNAP technology to autoconfigure the branch office router to provide call processing for Cisco
Unified IP phones that are registered with the router.
Public E.164
Cisco Unifed
"£L* j£- Calls ~*q|S
Communications
Manager
The Cisco Unified SRST gateway uses the local PSI'N breakout. Cisco Unified SRST features,
such as call preservation, autoprov isioning. and failover are supported.
During a WAN connection failure, when Cisco Unified SRST is enabled, Cisco Unified IP
phones display a message informing users that ihe phone is operating in Cisco Unified
Communications Manager fallback mode. This message can be adjusted.
While in Cisco Unified Communications Manager fallback mode, Cisco Unified IP phones
continue to send out keepalive messages to attempt to re-establish a connection with Cisco
Unified Communications Manager at the main site.
2-12 implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 >2010 Cisco Systems, Inc
Cisco Unified SRST Function: Switchback
Cisco Unified IP phones attempt to re-establish aconnection with Cisco Unified
Communications Manager at the main site periodically when they are registered with aCisco
Unified SRST gateway.
The default time that Cisco Unified IP phones wait before attempting to re-establish a
connection to aremote Cisco Unified Communications Manager is generally up to 120
seconds.
When the WAN link or connection to the primary Cisco Unified Communications Manager is
restored, after aconfigured waiting behavior, the Cisco Unified IP phones reregister with their
primary Cisco Unified Communications Manager. Three switchback methods are available on
the Cisco IOS router: immediate switchback, graceful switchback (after all outgoing calls on
the gatewav are completed), or switchback after aconfigured delay. Once switchback is
completed.'call processing reverts to the primary Cisco Unified Communications Manager, and
SRST returns to standby mode.
i Aopror, 60 Sec
E Time for SRST Phone regrsters with SRST ruuler
j Registration Process *i SRST router pulls IPphone coniiguration
I 10-20 Sec •*j Phone fully associatedwith SRST rouler
TCP Keepalive
: WANconnection restored phone
L Detault 30 Sec
—ti re establishesTCPconnector keepalive
Switchback Timer j from Unified CM* received agan
Default 120 Sec
Ifthe IP phone has an active standby connection that is established with a Cisco Unilied SRST
router, the fallback process takes 10 to 20 seconds after the connection with Cisco Unified
Communications Manager is lost. An active standby connection to a Cisco Unified SRS'f
router exists onlv- ifthe phone has a single Cisco Unified Communications Manager in its Cisco
Unified Communications Manager group. Otherwise, the phone activates a standby connection
to its secondary Cisco Unified Communications Manager.
Note The time that it takes for an IP phone to fall back to the Cisco Unified SRST router can vary
depending on the phone type Phones suchas the Cisco Unified IPPhone 7902G, 7905G.
and 7912G models can take approximately 2 5 minutes to fatl back to SRST mode.
Ifa Cisco Unified IP phone has multiple Cisco Unified Communications Manager systems in
its Cisco Unified Communications Manager group. Ihe phone progresses through its list before
attempting to connect with its local Cisco Unified SRST router. Therefore, the time that passes
before the Cisco Unified IPphone eventually establishes a connection with the Cisco Unified
SRST router increases witheach attempt to contact to a Cisco Unified Communications
Manager. Assuming that each attempt to connect to aCisco Unified Communications Manager
takes about 1minute, the Cisco Unified IPphone inquestion could remain offline for 3 minutes
or more following a WAN link failure. You can reduce this time by setting the keepalive timer
toa smaller value. You can configure the keepalive timer by using the Cisco CallManager
service parameter Station Keepalive Interval.
While in SRST mode. Cisco Unified IP phones periodically attempt to re-establish a connection
with Cisco Unified Communications Manager at the main site. The defaulttime that Cisco
Unified IP phones wait before attempting to re-establish a connection to Cisco Unified
Communications Manager is generally 120 seconds.
2-14 Implementing CiscoUnified Communications Manager, Part 2 (CIPT2) v8.0 )2010 Cisco Systems. Inc
MGCP Fallback Operation
This topic describes howthe MGCP gateway fallback mechanism works.
Cisco Unified
Communications
Manager
MGCP gateway fallback is a feature that improves the reliability of MGCP branch networks. A
WAN link connects the MGCP gateway at a remote site to the Cisco Communications Manager
at a central site, which is the MGCP call agent. If the WAN link fails, the fallback feature keeps
the gateway working as an H.323 or SIP gateway and re-homes back to the MGCP call agent
when the WAN link is active again. MGCP gateway fallback works along with the Cisco
Unified SRST feature.
Cisco IOS gateways can maintain links to up to two backup Cisco Unified Communications
Manager servers in addition to a primary Cisco Unified Communications Manager. This
redundancy enables a voice gateway to switch over to a backup server if the gateway loses
communication with the primary server. The secondary backup server takes control ofthe
devices that are registered with the primary Cisco Unified Communications Manager. The
tertiary backup takes control ofthe registered devices if both the primary and secondary backup
Cisco Unified Communications Manager systems fail. The gateway preserves existing
connections during a switchover to a backup Cisco Unified Communications Manager.
When the primary Cisco Unified Communications Manager server becomes available again,
control reverts to that server. Reverting to the primary server can occur in several ways:
immediately, after a configurable amount of time, or only when all connected sessions are
released.
If the WAN link (ails. MGCP gateways lose contact with Cisco Unified
Communications Manager.
MGCP gateway tnes to connect to the fallback Cisco Unified
Communications Manager.
MGCP gateway falls back to its default application (H.323 or SIP).
Cisco Unified
Communicati
Manager
2-16 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 ©2010 Cisco Systems. Inc
MGCP Gateway Fallback: Switchback
The switchback or re-home mechanism is triggered by the re-establishment ofthe TCP
connection between Cisco Unified Communications Manager and the Cisco MGCP gateway.
When the WAN link is restored, MGCP gateway starts the re-home function.
MGCP gateway reregisters with Cisco Unified Communications Manager.
Gateway switches back lo nomial MGCP application mode.
Cisco Unified
Communications
Manager
Re-home function in gateway-fallback mode detects the restoration of a WAN TCP connection
to the primary Cisco Unified Communications Manager server. When the fallback mode is in
effect, the affected MGCP gateway repeatedly tries to open a TCP connection to a Cisco
Unified Communications Manager server that is included in the prioritized list of call agents.
This process continues until a Cisco Unified Communications Manager server in the prioritized
list responds. The TCP open request from the MGCP gateway is recognized, and the gateway
reverts to MGCP mode. The gateway sends a RestartlnProgress (RSIP) message to begin
registration with the responding Cisco Unified Communications Manager.
All currently active calls that are initiated and set up during the fallback period are maintained
by the default H.323 session application, except ISDN Tl and EI PRI calls. Transient calls are
released. After re-home occurs, the new Cisco Unified Communications Manager assumes
responsibility for controlling new IP telephony activity.
MGCP Gateway
! He- .si;
-j -.. •rv-.,-. alien:, yar-j :••;
TCP Keepalive Detault 15 Sec
J WAN connection fails
M—
: After two missed keepalive messages.
; .->'niu.>','jli(,rri MdrSs!'. •
If the active Cisco Unified Communications Manager server fails to acknowledge receipt ofthe
keepalive message within 30 seconds, the gateway attempts to switch over lo thenextavailable
Cisco Unified Communications Manager server.
If none ofthe Cisco Linified Communications Manager servers responds, the gateway switches
into fallback mode and reverts to the default H.323 session application tor basic call control.
H.323 is a standardized communication protocol that enables dissimilar devices to
communicate with each other by using a common set of codecs, call setup and negotiating
procedures, and basic data-transport methods. Thegateway processes calls on itsown using
H.323 until one ofthe Cisco Unified Communications Manager connections is restored.
2-18 ImplementingCisco Unitied Communications Manager, Part 2 (CIPT2)vB.O ) 2010 Cisco Systems, Inc.
Cisco Unified SRST Versions and Feature
Support
This topic describes Cisco Unified SRST versions, their protocol support and features and the
required Cisco IOS Software release.
Extension
Mobiity
Eight active
cats per line ^ (new in 8.0)
Support for
E.164 numbers "' (new in ad)
with + prefix
Five additional
MOH streams (new in 8.0)
(SCCP only)
The version ofthe Cisco Unified SRST application depends on the release ofthe Cisco IOS
Software that is running on the router. Each Cisco IOS Software release implements one
particular Cisco Unified SRST version. You can upgrade to anewer version ofCisco Unified
SRST via aCisco IOS update. Some ofthe recent Cisco IOS Software releases have higher
memory requirements than older releases, so make sure that you consider these requirements
before upgrading. n
For detailed information about Cisco Unified SRST versions and their hardware and feature
support, refer to the Cisco Unified Survivable Remote Site Telephony Version 8.0 data sheet"
http://w^vv.cisco.com/cn/LlS/prod/collatcraI/voicesw/ps6788/vcallcon/ps2l69/
data_shect c78-57048i.html.
800 Series 4
1861 15
2801-2851 25-100
2901-2951 35-250
3925-394 5E 730-1500
The figure shows asummarv ofthe maximum number ofphones that Cisco Umtied SRST
routers can accommodate. For more details, such as minimum memory requirements, refer to
-Cisco Unified SRST 8.0 Supported Firmware. Platforms, Memory, and Voice Products
(imp'/.'vvwvvci^co.coni.ctTU'S./docs/voice ipeomm/cusrst/rcquircmcnts/guidc/srsXOspc.himl)
These maximum numbers of IP phones are for common Cisco Unified SRST configurations
Note
only Systems with large numbers of IP phones and complex configurations may not work on
all platforms and can require additional memory or ahigher performance platform.
Summary
References
For additional infonnation. refer to these resources:
• Cisco Sv stems. Inc. Cisco Unified Communications System 8.x SRND, April 2010.
http:/ www.cisco.com'cn'l 'S/does/voice ip comni/'cucm/smd/oVucSvlitmi
• Cisco Svstems. Inc. Cisco UnifiedCommunications ManagerAdministration Guide
Release 8.0(1>. Februarv 2010.
http:/';vuv wxisco.com/cn/1 :S.;docs.'\oicc_iji_coinm/cucm/admin/S I) l/ccmefg/hccm-80!-
cin.litinl
• Cisco Systems. Inc. Cisco IOS Voice Configuration Library (with Cisco IOS Release 15.0
updates}. July 2007.
hltp:/v\v\v\.ciM.'o.coni,'cn.'US/'docs/iosT2 3/vvf_c/cisco_ios voice_conf!giii'ation_librarv_t
iossarv \cl.htm
Choose previously
configured gatekeeper.
Finally, you need to provide the gatekeeper information. From the drop-down list, choose the
gatekeeper that this trunk should register to, and then choose the terminal type. Cisco Unified
Communications Manager can register trunks as terminals or gateways with an H.323
gatekeeper. Usually die terminal type is set to Gateway.
In the Technology Prefix field, enter the prefix, which should be registered with the gatekeeper.
Note The prefix that you enter is the prefix that the trunk will register with the gatekeeper. It can,
but does not have to, include a technology prefix. In the example, a prefix of 408 is used.
More information about prefixes and technology prefixes is provided in the Implementing
Cisco Voice Communications and QoS (CVOICE) course.
Tip The H.323 zone name is case-sensitive. Make sure that it matches the zone name that has
been configured at the gatekeeper.
1-104 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) vS.O >2010 Cisco Systems, Inc.
Unified CM* Gatekeeper-Controlled ICT
and H.225 Trunk Configuration (Cont.)
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After vou have configured the gatekeeper, you can add the gatekeeper-controlled trunk.
Navigate to Dc\ice > Trunk and click Add New. fhen choose the trunk type. As discussed
earlier in this lesson, there are two tvpes of gatekeeper-controlled 11.323 trunks: gatekccpei-
controlled ICTs (which vou have to use when connecting to a version of Cisco CallManager
earlier than version 3.2) or 11.225 trunks (which are used to connect to Cisco Unified
Communications Manager Version 3.2 or later, as well as other H.323 devices such as
gateways or conferencing svstems).
After selecting the trunk tvpe. enter a name and description for the trunk and choose the device
pool that should be used.
Enter description.
Make sure
gatekeeper is
enabled.
1-102 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 ) 2010 Cisco Systems, Inc.
Cisco Unified Communications
Manager Nongatekeeper-Controtled
ICT Configuration (Cont.)
^'•^•msammmtm
• tritlsl^-.l-.,!™^
Enter IP address
Then enter the II* address or addresses ofthe Cisco Unified Communications Managerservers
ofthe other cluster.
Note Because the nongatekeeper-control led ICT does not use a gatekeeper for address
resolution, you must manually enter the IP addresses of the devices on the other side.
The figure shows how to add a new nongatekeeper-controlled ICT. First, you navigate to
Device > Trunk-and then click Add New.
Next, you must choose the appropriate trunk type. After you click Next, the Trunk
Configuration window appears, where you can configure the nongatekeeper-controlled ICT.
Enter a device name and description, and choose the device pool that should be used.
1-100 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) vS.O >2010 Cisco Systems, Inc.
Cisco Unified Communications
SIP Trunk Configuration (Cont.)
SIP trunk security profiles are used to enable and disable security features on
SIP trunks; you configure them by navigatingto System > Security Profile >
SIP Trunk Security Profile Adefauft profile (with security disabled) exists.
SIP profiles are used to set timers and some feature settings. You configure
them by navigating to Device > Device Settings > SIP Profile. A default profile
exists
Enter IP address of
other device at end
of SIPtrimk
In the SIP Information area of the Trunk Configuration window, enter the destination addr:ss.
This IP address is for the dev ice that is located on the other end ofthe SIP trunk. This device
can be a Cisco Unified Border Element, Cisco Unified Communications Manager Express, or
any other SlP-capable device, such as a third-party SIP proxy server.
In addition, you must choose a SIP trunk security profile and a SIP profile. Both parameters are
mandator, and do not have a default value.
The SIP trunk security profile is used to enable and configure security features on SIP trunks,
such as Transport Layer Security (TLS) with two-way certificate exchange, or SIP digest
authentication. One default SIP trunk security profile exists: the nonsecure SIP trunk profile,
which has security disabled. You can configure additional SIP trunk security profiles by
nav igating to System > Security Profile > SIP Trunk Security Profile.
The SIP profile is used to set timers, Real-Time Transport Protocol (RIP) port numbers, and
some feature settings (such as Call Pickup Uniform Resource Identifiers [URIs], call hold
ringback. or caller ID blocking). One default SIP profile exists: It is called a standard SIP
profile. You can configure additional SIP profiles by navigating to Device > Device Settings >
SIP Profile.
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To add a SIP trunk in Cisco Unified Communications Manager, navigate to Device > Trunk
and click Add New. Then, in the Trunk Type drop-down list, choose SIP Trunk and click
Next.
In the Trunk Configuration window, enter a name and description for the SIP tmnk and choose
the device pool that should be used.
1-98 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 ©2010 Cisco Systems, Inc.
Trunk Types Used by Special Applications
This subtopic describes additional trunk types that can be configured in Cisco Unified
Communications Manager.
Note More information about Cisco EMCC and SAF trunks will be provided in the corresponding
lessons of this course.
The trunk (which points to the gatekeeper), the route group, the route list, and the route pattern
configuration are the elements ofthe gatekeeper in which you have to specify the IP address of
the gatekeeper. This implementation is like the implementation ofagateway.
1-96 Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 © 2010 Cisco Systems. Inc.
Trunk Implementation Overview
This topic describes how to configure trunks in Cisco Unified Communications Manager.
Cisco Unified
Communica
Manager Access and Site
Cluster Code. 9 222
4- Digit
Directory
Numbers
Nongatekeeper-controlled ICT
and SIP trunk configuration:
Cisco Unified - Trunk with IP address of peer
Communicalions
• Route pattern, route list, route group
Manage
Cluster
The figure illustrates the most important configuration elements for implementing a SIP or
nongatekeeper-controlled ICT in Cisco Unified Communications Manager, "fhese elements arc
the configuration ofdie trunk itself, in which you have to specify the IP address ofthe peer, as
well as the route group, route list, and route pattern configuration, fhis implementation is like
the implementation of a gatewav.
ImplementingMultisite Deployments
'2010 Cisco Systems. Inc
Cisco IOS H.323 Gateway Configuration
This subtopic describes how to configure a Cisco IOS router as an H.323 gateway.
When configuring an H.323 gateway, the first task is to enable H.323 at one IP interface. If
multiple IP interfaces are present it is recommended that you use a loopback interface.
Otherwise, if the interface that has been selected for H.323 is down, the H.323 application will
not work, even if-other interfaces could be used to route the IP packets. In this example, there is
only one Ethernet interface, and H.323 has been enabled on that interface, using the h323-
gateway voip interface and h323 gateway voip bind srcaddr IP address commands.
In contrast to MGCP gateways in which the call agent takes care of call routing, H.323
gateways require local dial plan configuration. In the example, the H.323 gateway is configured
with a VoIP dial peer that routes calls that are placed to the PSTN number 511555... ofthe
gateway toward Cisco Unified Communications Manager. The gateway receives these calls
from the PSTN because 511555 1001-1003 is the direct inward dialing (DID) range ofthe
PSTN interface (port 0/0/0:23). In addition, the PSTN gateway is configured with a POTS dial
peer that routes all calls starting with 9 out to the PSTN, using the ISDN PRI (port 0/0/0:23).
Note that the configured digits of a destination pattern in a POTS dial peer are automatically
stripped off. Therefore, the 9 is not sent out to the PSTN. In the other direction, the gateway
does not perform any digit manipulation because VoIP dial peers do not strip off any digits
automatically. Cisco Unified Communications Manager receives H.323 call setup messages for
calls that were received from the PSTN in their entire length (usually 10 digits). Because the
internal directorv numbers arc four digits, either Cisco Unified Communications Manager or
the H.323gatewav need to be configured to strip the leadingdigits so that the remaining four
digits can be used to route the call to internal directory numbers.
Note More informationon how to implement digit manipulation is provided in the next lesson of
this module.
1-94 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 J2010 Cisco Systems, Inc.
Cisco Unified Communications Manager H.323 Gateway
Configuration
This subtopic describes how to configure an 11.323 gateway in Cisco Unified Communications
Manager.
-""•- -
-•:•• zm
'JE-.ll.;rt..t,Lr >•:•>•,
To add an H.323 gateway to Cisco Unified Communications Manager, navigate to Device >
Gateway and click Add New. fhen, from the Gateway Type drop-down list, choose H.323 and
click Next.
In the tiateway Configuration window, enter the IP address ofthe 11.323 gateway in the Device
Name field, enter a description, and select the device pool that should be used. If Cisco Unified
Communications Manager should consider only some ofthe called digits, you can set the
significant digits parameter to the numberof least significant digits that shouldbe used for
routing inbound calls. In the example that is provided in the previous topic, in which the
gateway sends complete 10-digit PSTN numbers to Cisco Unified Communications Manager,
setting the significant digits to 4 wouldallow the incoming calls lo be routed to internal
director; numbers without any additional configuration (such as translation patterns).
1-92 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) vS.O © 2010 Cisco Systems, Inc.
H.323 Gateway Implementation Review
This topic describes how to implement an H.323 gateway.
1001-1003
To implement an H.323 gatewav. you first must add the gatewav to Cisco Unified
Communications Manager. When adding the gateway, you need lo specify the IP address ofthe
gateway.
Note More information about the H.323 protocol and H.323 gateway characteristics have been
provided in the ImplementingCisco Voice Communications and QoS (CVOICE) course.
MGCP gateway implementation with Cisco Unified Communications Manager has been
covered in detail in the Implementing Cisco Unified Communications Manager Part 1
(CIPT1) course This topic is only a high-level review of H 323 gateway implementation.
Then vou need to configure the Cisco IOS gateway by following these steps:
Step 1 Configure the H.323 gateway, specifying its H.323 ID and the IP address to use.
You do this configuration on any interface, typically on a loopback interface, Fnsure
that you use the same IP address that you configured in Cisco Unified
Communications Manager for the H.323 gateway.
Note Ifthe IP address that is configured in Cisco Unified Communications Manager does not
match the IP address that is used by the gateway, Cisco Unified Communications Manager
considers the H 323 signaling messages to be sent from an invalid (unknown) source and
ignores them. However, itdoes not ignore the messages ifpromiscuousoperation has been
permitted (thisservice parameter can be configured in Cisco Unified Communications
Manager)
1-90 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0 © 2010 Cisco Systems, Inc.
Configuring Cisco IOS Gateway for MGCP—Example
The figure shows an example of a Cisco IOS MGCP gateway that pulls its configuration from a
configuration server.
In the example, there is one Cisco Unified Communications Manager server (providing call
processing and TFTP services) with the IP address 10.1.1.1. 'fhere is a Cisco IOS MGCP
gateway with a connection to the PSTN using an Fl interface (port 0/1/0). The gateway and its
HI PRI endpoint have been added to Cisco Unified Communications Manager. At the gatewav.
the commands ccm-manager config server 10.1.1.1 and ccm-manager config server have
been entered. No MGCP configuration commands have been manually entered, because the
MGCP configuration is automatically downloaded and applied by the configuration server
feature.
After the gatewav downloaded its cnf.xml configuration file from the Cisco Unified
Communications Manager TFTP server, these MGCP commands were added and saved to
NVRAM:
controller SI 0/1/0
framing crc4
linecode hdb3
interface Serial0/l/0:15
isdn switch-type primary-4ess
isdn incoming-voice voice
isdn bind-13 ccm-manager
i
ccm-manager mgcp
ccm-manager music-on-hold
Note More information about manual configuration of MGCP gatewaysis provided inthe CVOICE
course.
Note Be aware that, as long as the configuration server isactive onthe Cisco IOS gateway, every
time the MGCP endpoint is resetfrom Cisco Unified Communications Manager, the Cisco
IOS configuration also will be rewritten. In addition, when you reload theMGCP gateway,
the MGCP configuration will be rewritten as long as the configuration server is enabled.
L
Therefore, itis common practice to use the configuration only for initial configuration when
manual changesare required. After you modify the downloaded configuration, you M
deactivate the configuration serverso thatthe manually addedchangesare preserved '••»
m,
First, reset the MGCP gatewayor MGCP endpoint inCisco Unified Communications
Manager. Then enterthe no mgcp command, followed by the mgcp command in
configuration at the Cisco IOS gateway.
L
I
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 ©2010 Cisco Systems, Inc
Cisco IOS Gateway MGCP Configuration Methods Review
This subtopic reviews how to configure a Cisco IOS MGCP gateway to integrate with Cisco
Unified Communications Manager.
Ailer adding the MGCP gatewav in the Cisco Unified Communications Manager web
administration, vou need to configure the Cisco IOS MGCP gateway to register ittothe Cisco
Unified Communications Manager, 'fhere are three methods for configuring a Cisco IOS
Software-based gateway to register itto Cisco Unified Communications Manager via MGCP:
• Cisco IOS MGCP gateway configuration with theuse of a configuration server:
— Specifv the IP address ofthe configuration server (Cisco Unified Communications
Manager TFTP server).
— If more than one Cisco Unified Communications Manager TFTP server is deployed
inthe Cisco Unified Communications Manager cluster, configure the gateway with
all Cisco Unified Communications Manager TFTP server IP addresses.
— Fnable the configuration server feature.
• Manual Cisco IOS MGCP gateway configuration:
— Specify the IP address ofthe MGCP call agent (Cisco Unified Communications
Manager server).
— Ifmore than one Cisco Unified Communications Manager server is used for call
processing (that is. running the Cisco CallManager service), configure the gateway
with aprimary and redundant call agent by specifying the IP addresses oftwo Cisco
Unified Communications Manager call-processing servers.
— Configure global MGCP parameters.
Examples ofglobal MGCP configuration commands arc mgcp packet and mgcp
rtp commands.
L
To implement an MGCP gateway, you first need to add the gateway to Cisco Unified
Communications Manager. Next, you add voice modules and voice interface cards (VICs) to L
the gatewav. and finally, you configure the endpoints.
Note More informationabout the MGCP and MGCP gateway characteristics are provided in the L
ImplementingCisco Voice Communications and QoS (CVOICE) course. MGCP gateway
implementation with Cisco Unified Communications Manager has been covered in detail in
Implementing Cisco Unitied Communications Manager, Part 1 (CIPT1) course. This topic is
only a high-level review of MGCP gateway implementation.
L
After adding the MGCP gateway and its endpointsand configuring the endpoints in Cisco
Unified Communications Manager, you need to configure the MGCP gateway itself. Cisco L
Unified Communications Manager stores at its TFTPserver an XMLconfiguration file that can
be downloaded by the MGCP gateway. Alternatively, you can configure the gateway manually.
L
1-86 Implementing Cisco Unified Communications Manager, Part 2 (C1PT2) v8.0 )2010 Cisco Systems, Inc.
With a gatekeeper-controlled ICT. you configure only one trunk. That trunk then communicates
via the gatekeeper with all other clusters that are registered to the gatekeeper. If a cluster or
subscriber becomes unreachable, the gatekeeper automatically directs the call to another
subscriber in the cluster or rejects the call if no other possibilities exist, fhis action allows the
call to be rerouted over the PSTN (if required) with little incurred delay. With a single Cisco
gatekeeper, it is possible to have 100 clusters that arc registering a single trunk each, with all
clusters able to call each other. With nongatekeeper-controlled trunks, this same topology
would require 99 trunks to be configured in each cluster. The gatekeeper-controlled ICT should
be used for communicating only with other Cisco Unified Communications Managers,because
the use of this trunk with other 11.323 devices might cause problems with supplementary
services. In addition, a gatekeeper-controlled ICT must be used for backward compatibility
with Cisco Unified Communications Manager versions earlier than Version 3.2 (referred lo as
Cisco CallManager).
Ihe H.225 trunk is essentially the same as the gatekeeper-controlled IC'1. except that't has the
capability of working with Cisco Unified Communications Manager clusters (Version 3.2 and
later), as well as other 11.323 devices, such as Cisco IOS gateways (including Cisco Unified
Communications Manager Express), conferencing systems, and clients. This capability is
achieved through a discovery mechanism on a call-by-call basis. This type of trunk is the
recommended H.323 tmnk if all Cisco Unified Communications Manager clusters are at least
Version 3.2.
Nongaiefteeper- Gatekeeper-
ControlledlCT ControBedlCT
Cisco Unified
Peer Communications
Manager
eJS^jESiSSS
The nongatekeeper-controlled ICT isthe simplest, since itdoes not use a gatekeeper. Itrequires
the IPaddress ofthe remote Cisco Unified Communications Manager server or servers to be
specified, because the dialed number isnot resolved toan IP address by a gatekeeper. Call
Admission Control (CAC) can be implemented by locations but not by gatekeeper CAC.
Scalability is limited because no address resolution is used andall IP addresses have to be
configured manually. The nongatekeeper-controlled ICT points to theCisco Unified
Communications Manager server ofthe other cluster.
You may define up to three remote Cisco Unified Communications Manager servers in the
same destination cluster. The trunk will automatically load-balance across all defined remote
Cisco Unified Communications Manager servers. In the remote cluster, it is important to
configure a corresponding ICT (nongatekeeper-controlled) thathas a Cisco Unified
Communications Manager group containing the same servers that were defined as remote Cisco
Unified Communications Manager servers in the first cluster. Asimilar configuration is
required in each Cisco Unified Communications Manager cluster that isconnected by the ICTs.
Fora larger number of clusters, the gatekeeper-controlled ICT should be used instead ofthe
nongatekeeper-controlled trunk. The advantages of using the gatekeeper-controlled tmnk are
mainly the overall administration ofthe cluster and failover times. Nongatekeeper-controlled
trunks generally require that a full mesh oftrunks be configured, which can become an
administrative burden asthe number ofclusters increases. In addition, ifa subscriber server ina
cluster becomes unreachable, there will be a 5-second (default) timeout while the call is
attempted. Ifan entire cluster isunreachable, the number ofattempts before either a call failure
ora rerouting ofthe call over the PSTN will depend onthe number of remote servers that are
defined for the tmnk and on the number oftrunks in the route list or route group. Ifthere are
many remote servers and many nongatekeeper-controlled trunks, the call delay can become
excessive.
1-84 Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 ©2010 Cisco Systems, Inc.
H.323 Trunk Overview
The figure illustrates varioustypes olT1.323 trunks.
Cisco Unified
Communications
Manager
Cluster A
Cisco Unified
Com mu nica tions
Manager
Cluster D
In the example, the Cisco linified Communications Manager cluster Auses anongatekeeper-
controlled ICT to Cisco Unified Communications Manager cluster B. In addition. Cisco
Unified Communications Manager cluster A isconfigured with a gatekeeper-controlled ICT .
The gatekeeper-controlled ICf points to agatekeeper, which is used for address resolution. In
this example, the gatekeeper can route calls between Cisco Unified Communications Manager
clusters A. C. and D.
Main
Site
SIP uses the distributed call-processing model, so a SIP gateway or proxy has its own local dial
plan and performs call processing on its own. A Cisco Unified Communications Manager SIP
tmnk can connect to Cisco IOS gateways, a Cisco Unified Border Element, other Cisco Unified
Communications Manager clusters, or a SIP implementation with network servers (such as a
SIP proxy).
SIP is a simple, customizable protocol with a rapidly evolving feature set.
Note When you use SIP trunks, Media Termination Points (MTPs) might be required if the
endpoints cannot agree on a common method of dual tone multifrequency (DTMF)
exchange.
1-82 ImplementingCisco Unified Communications Manager, Part 2 (CIPT2)vS.O >2010 Cisco Systems, Inc.
Cisco IOS Gateway Protocol Comparison Review
fhe figure reviews the advantages and disadvantages of 11.323 gateways. MGCP-controIlcd
gateways, and SIP gateways.
fach ofthe three gateway protocols has advantages and disadvantages when compared with
each other. There is no generally "best" gateway protocol. You should select the most
appropriate protocol, depending on the individual needs and demands in a Cisco Unified
Communications Manager environment.
Note The Implementing Cisco Voice Communications and QoS (CVOICE) course provides
detailed information on functions and features ofthe H.323, MGCP, and SIP.
Call applications
No Yes Yes
usable
I' Support irtroQuced with Cisco Unrfied Communications Manager Version 8.0 |
As shown in the table, the three main gateway signaling protocols—MGCP, H.323, and SIP—
provide various features and functions when implemented with Cisco Unified Communications
Manager and Cisco IOS gateways.
1-80 Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) vS.O >2010 Cisco Systems, Inc.
Cisco Unified Communications Manager Connection Options
Overview j
In Cisco Unified Communications Manager, you can configure gateways and trunks for
connections to the public switched telephone network (PSTN) or to other VoIP domains.
• Gateways
- H.323 (any H 323 device when not using a gatekeeper)
- MGCP
H.323
• Direct connection to another Cisco Unified
Communications Manager cluster
• Connection to any H.323 device via a gatekeeper
SIP
Gateways areconfigured bv the VoIP protocol (hat they use. Cisco Unified Communications
Manager supports H.323 gateways. Media Gateway Control Protocol (MGCP) gatewavs. and
Skinny Client Control Protocol (SCCP) gatewavs. Trunks can be configured as H.323 trunks
(three types are available) or SIP trunks.
Trunks and gatewavs arcconfigured when connecting to devices that allow access to multiple
endpoints. If the destination is a single endpoint. phones areconfigured. Phones can be
configured as SCCP. SIP. or 11.323.
When Cisco Unified Communications Manager routes calls lo a device that is using MGCP.
SCCP. or SIP. it is obvious which type of deviceto add. becausethese protocols can be
configured onlv with either a gateway or a trunk. Inthe case of 11.323. however, an H.323
gateway as well as an H.323 trunk can be configured, and it is important to know whether to
use the gateway or the trunk. You use H.323 trunks only when connecting toanother Cisco
Unified Communications Managerserver (eithera clusteror a standalone Cisco Unified
Communications Manager server, in the caseof Cisco Unified Communications Manager
Business fdition) or when using an H.323 gatekeeper.
H.323 gatewavs are configured when connecting to any other H.323 device that isnot an
endpoint. Such dev ices can be Cisco IOS H.323 gateways or11.323 gateways ofother vendors.
Main
Site
The figure shows a Cisco Unified Communications Manager cluster at the main site with these
connections to other sites:
1-78 Implementing Cisco Unified Communications Manager, Part2 (CIPT2) v8.0 12010 Cisco Systems, tnc.
Lesson 3
Implementing Multisite
Connections
Overview
Cisco Unified Communications Manager multisite deployments can use various connection
options between sites. Ihis lesson describes connection options and explains how toconfigure
them.
Objectives
Upon completing this lesson, you will be able to configure gateways and trunks in multisite
nvironmenls. This ability includes being able to meet these objectives:
Identify the characteristics ofthe trunk and gateway types that aresupported by Cisco
Unified Communications Manager
Describe how lo implement MGCP gateways
Describe bow to implement 11.323 gateways
Describe various types of trunks that aresupported by Cisco Unified Communications
Manager
Describe how to implement SIPtrunks in Cisco Unitied Communications Manager
Describe how to implemeni intercluster and H.225 trunks in Cisco Unitied
Communications Manager
1-76 Implementing Cisco Unified Communications Manager, Part2 (CIPT2) vS.O © 2010 Cisco Systems; Inc.
References
for additional information, refer to these resources:
• Cisco Svstems. Inc. Cisco Unified Communicalions System 8.x SRND, April 2010.
lutp: Asww.cisco.com/en/US/docs.voice ip comm/cucm/^rnd/8\/uc8\.h(inl
• Cisco Svstems. Inc. ('isco (Unified Communications Manager Administration Guide
Release 8.0(1). February 2010.
Imp: 'wwu,ciscn.com/eiiT,S/doev\oice_ip_comm/cucm/admin/8 OJ/cemciU/bccm-801
cm.him!
Summary
Summary (Cont.)
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Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
'2010 Cisco Systems. Inc.
Cisco Unified Border Element in Flow-Through Mode
The figure illustrates how the use of aCisco Unified Border Element protects inside devices
such afasco Unified Communications Manager and IP phones by acting as asignaling and
media proxy.
Private IP Public IP
10.2.1.5 Address: AddressA
Private IP Network' 1000.0/8 10.3.1.1
..1.1.110 10 31.1
Signaling: 10.
|~| Signaling: A(Public IP) to B(Public IP) |
RTP-1Q2.1.5.O10311 — RTP: A(Public IP) to B(Public IP)
In the example. Cisco Unilied Communications Manager has aprivate IP addrcs of 10.1.1.1.
and the IP phone has aprivate IP address of 10.2.1.5. ACisco Unified Border Element
o, t the Cisco Unified Communications Manager cluster to the outside world, in this case.
^ ninternet telephony service provider (ITSP,. The Cisco Uinfie Border hcment is
configured in flou-through mode and uses an internal private IP address ot 10... Iand an
external public IP address of A.
When Cisco Unified Communications Manager wants to signal calls to the ITSP. it does not
Tend he packets to the IP address ofthe ITSP (IP address D). Instead it sends them to the
ternalkddressoHhe^
configuration Cisco Unified Border Element then establishes asecond cal leg to the 1ISP.
unCi blic IP address Aas the source and IP address B(ITSP) as the desUr.Uo^Occ the
call is set up the Cisco Unified Border Element terminates Rl Ptoward the ITSP. using its
pX II' address, and sends the received RTP packets to the internal IP phone, using its
internal IP address.
This solution allows Cisco Unified Communications Manager and IP phones to communicate
on v htne ntemal. priv ate IP address ofthe Cisco Unified Border Element, he onlv IP
dre h sible to the ITSP is the public IP address ofCisco Unified Border f.lcment.
When Ctsco Unified Communications Manager servers and IP phones need to connect lo the
Internet. Cisco Unified Border Element can be used as an application proxy. When used in this
way, Cisco Unified Border Element splits off-net calls inside and outside into two separate call
,,'.-!f° °rder Element also features signaling interworking from SIP to SIP SIP
to 11.323, H.323 to SIP. and H.323 to H.323.
The Cisco Unified Border Element can function in two modes:
• Flow-around: In this mode, only signaling is intercepted by Cisco Unified Border
Element Media exchange occurs directly between endpoints (and>*M around Cisco
Unified Border Element). Only signaling devices (Cisco Unified Communications
Manager) are hidden from the outside.
' ,Fi°r"?M0U?h: In thiS m0dc" both siSnalinS **media slr<*ms are intercepted by Cisco
Unified Border Element (byflowing through Cisco Unified Border Element). Both Cisco
Unified Communications Manager and IP phones are hidden from the outside.
In flow-through mode, only Cisco Unified Border Element needs to have apublic IP address
so NAT and security issues for internal devices (Cisco Unified Communicalions Manager '
ouZe it should
outside, it sh tTu 7 I"'-against
be hardened B£CaUSe CiSC° Unified B0nler Hlement is "P""" ^the
attacks.
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Implementing Cisco Unitied Communications Manager, Part 2(CIPT2) vS.O
©2010 Cisco Systems. Inc.
Localization of callingparty numbers a. phones: Caller IDs sPl yed at phon n
also be localized so that the end users are not lim.ted to seeing all callers ,n W»A™*
fonnat Again, global transfonnations (ofthe calling-party number only in this case) can
Tsed sothat caller 1Ds. which might be different at each site. -™^^^'^
format Similarh. the globalized calling-party number is also maintained in call lists so that
users can place callbacks to globalized numbers without needing lo edit the number.
Substantial simplification of dial plans: With local route groups and global
transfonnations. globalized call routing drastically reduces the s™^™^™%£*ld]
plans Features such as TEHO. AAK. SKST. CEUR. Cisco Device Mobil, y. and Cisco
Extension Mobilitv can be implemented much more easily in international deployments.
1-71
Multisite Deployment Implementation
© 2010 Cisco Systems. Inc
Globalized Call Routing Advantages
There are several advantages ofglobalized call routing that are especially applicable to
international multisite deployments.
It is evident from the figure that, in several situations, the numbers that are provided at call
ingress do not conform to the normalized format to be used for call routing. 1he same situation
occurs with cal! egress, where the normalized fonnat is not always used when the call is being
delivered. Therefore, localized call ingress has to be normalized (that is. globalized), and
globalized fonnat has to be localized at call egress.
On the left side ofthe figure, call ingress is illustrated by two types ofcall sources:
• External callers: Their calls are received by Cisco Unified Communications Manager
through agateway or tmnk. In aPSTN gateway, calling- and called-party number are
usuallv provided in localized E.164 format.
• Internal callers: Their calls are received from internal phones. Ifcalls to internal
destinations (for example, phone to phone), calling- and called-party numbers are typically
provided as internal directory numbers. Ifcalls to extemal destinations (for example, phone
to PSTN), the calling number is the directory number (at call ingress time) and the called
number depends onthe local dial rules for PSTN access. These dial rules can differ
significantly per location.
The center ofthe figure illustrates the standards that are defined for normalized call routing As
mentioned earlier, because most calls use global E.164 format, this process is also referred to as
globalized call routing. Here arethe defined standards:
• External to internal
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Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 ©2010 Cisco Systems, Inc
Globalized Call-Routing Overview
(Cont.)
Alter the call has been routed and path selection (ifapplicable) has been performed, the
destination dev ice might need to change the nomialized numbers lo local format. 1his situation
is referred to as localized cal! egress.
Localized call egress applies tothese kinds ofnumbers:
• Calling- and called-partv numbers for calls that are routed to gateways and trunks: If
the PSTN or the telephonv svstem on the other side of atrunk docs not support globalized
call routine the called- and calling-party numbers need lo be localized from global format.
An example would be to change the called-party number +494012345 to 011494012345
before sending the call out tothe PSTN in the United Slates.
• Calling-party numbers for calls that are routed from gateways or trunks to phones:
fhe nhone user mav want to see caller IDs in alocal format rather than aglobal format, for
example, auser at aU.S. phone may want to see PSTN callers who are located in the same
area code as 7- or 10-digil numbers and not with »1 followed by 10 digits.
1ocalized call egress is not needed for the called-party number ofcalls that are routed to
phones because internal director}' numbers are the standard (normalized) formal for internal
destinations (regardless ofthe source ofthe call). These numbers might have been dialed
ditTerentlv initially. In that case, however, this localized call ingress was normalized before call
routing.
Localized call egress is also not required for the calling-party number of internal calls (internal
to internal), because typically the standard for ihe calling-parly number ot such calls is lo use
internal directory numbers.
Note
When internal directory numbers are not unique (for example, when there are overlapping
directory numbers at various sites), the called- and calling-party numbers of internal calls
can be globalized at call mgress and localized at call egress just like external calls.
Note Except for the mentioned internal calls (where the destination is adirectory number and in
the case of an internal source, the source is a directory number), all numbers are normalized
to E.164 global format. Therefore, this call-routing implementation model is referred to as
globalized callrouting.
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implementing Cisco Unified Communications Manager, Pari 2(CIPT2) v8.0 " ©2010 Cisco Systems, Inc.
Globalized Call-Routing Overview
Globalized call routing simplifies the implementation of international Cisco Communications
Manager depknments.
Note Alt these elements have been discussed in other courses, such asImplementing Cisco
Voice Communications and QoS (CVOICE) and Implementing Cisco Unified
Communications Manager, Part 1(CIPT1). However, information onhow to use these
elements to implement a dial plan in multisite deployments is provided in a separate lesson
of this module.
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Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 '2010 Cisco Systems. Inc.
Least Cost Routing, tail-end hop-off (TEHO), and PSTN backup: Can be implemented
b> appropriate call routing and path selection that is based on priorities.
Globalized call routing: In this dial plan implementation, all received calls arc normalized
toward a standardized fonnal. fhe formal that isused in call routing isglobalized format,
because all numbers are represented in E. 164 format with a4 prefix, 'fhe process of
nomializine the numbers as dialed by end users (localized ingress) istherefore also referred
to as globaUzation. Once the localized input has been globalized during ingress, the call is
routed based on globalized numbers. After call routing and path selection, the called
number is localized during call egress, depending on the selected egress device.
Dial plan issues in multisite deployments can be solved in the following ways:
• Overlapping and nonconsecutive numbers: Solved by implementing access codes and
site codes tor intersite dialing. This approach allows call routing that is independent of
directorv numbers. Appropriate digit manipulation (removal ofsite codes in called number
of outgoing calls) and prefixing ofsite codes in calling number ofincoming calls are
required.
• Direct inward dialing (DID) ranges and E.164 addressing: Solutions for mapping of
internal directory numbers to PSTN numbers include DID, use ofattendants or interactive
voice response (IVR) applications to transfer calls, and extensions that are added to PSTN
numbers invariable-length numbering plans.
' 5ITT1
that ,s basedTmbT^reSeu!aIi0n
on TON enables the '"standardization
'SDN (,ype °fofnumbers
number'orthatT0N>: Di8jt manipulation
are signaled using
different TONs.
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Implementing Cisco Unified Communications Manager, Part 2(CIPT2) vS.O S2010 Cisco Systems, Inc.
Mobility Solutions
This subtopic provides an overv iew about mobility solutions that solve issues that are the result
ofroaming users and devices, and multiple telephones (office phone, cell phone, home phone,
and so on).
Mobility Solutions
When users or de\ ices roam between sites, issues arise that can be solved by these mobility
solutions:
• Cisco De* ice Mobilitv: Solves issues that arc caused by roaming devices, including
invalid device configuration settings such as regions, locations, SRST reference. AAR
groups, calling search spaces (CSSs). and so on. The Cisco Device Mobility feature of _
Cisco Unitied Communications Manager allows device settings that depend on the physical
location ofthe device to be automatically overwritten ifthe device appears in adifferent
phvsical location.
. Cisco Extension Mobilitv: Solves issues that are the result of roaming users using shared
guest IP phones that are located in other offices. Issues include wrong directory number,
missing IP Phone Services subscriptions. CSS. and so on. Cisco Extension Mobility allows
users to log in to guest phones and to replace the configuration ot the IP phone with the II
phone configuration ofthe logged-in user.
• Cisco I'nified Mobilitv: Solves issues of having multiple phones and consequently
multiple phone numbers, such as an office phone, cell phone, home (office) phone, and so
on. Cisco Unitied Mobility allows users to be reached by asingle number, regardless ot the
phone that isactually used.
Note Cisco Device Mobility and Cisco Extension Mobility will be discussed in detail in later
lessons of this course. Cisco Unified Mobility has been discussed in detail in the
Implementing Cisco Unified Communications Manager, Part 1(CIPT1) course
Cisco Unified
Communications
Manager
Ifa call over the IP WAN is not admitted by CAC, the call can be rerouted over the PSTN
using AAR. The AAR feature includes an option that allows the alternate number to be set per
IP phone. This option is also known as Call Forward No Bandwidth (CFNB) In the example
because the remote site does not have PSTN access, the call is not rerouted to the IP phone over
the PSTN (instead of over the IP WAN). It is alternately rerouted to the cell phone ofthe
affected user. AAR and CFNB improve availability in multisite environments by making it
possible to reroute on-net calls thatfailed CAC overthe PSTN.
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Implemenling Cisco Unrfied Communications Manager, Part 2(CIPT2) v8.0 ©2010 Cisco Systems, Inc.
Using CFUR to Reach Users of Unregistered Software IP
Phones on Their Cell Phones
This subtopic describes how CFUR can be used to route calls lo the cell phones ofusers who
have shut down PCs that have asoftphone installed.
_y
If amobile user has alaptop with asoftphone (for instance, Cisco IP Communicator) and shuts
down the laptop. CFUR can be used to forward calls placed to the softphone to the cell phone
of auser The user docs not have to set up Call Forward All (Cl'A) manually before closing the
softphone application. However, ifthe softphone is not registered, calls are forwarded to the
cell phone ofthe user. This action is another application ofthe CFUR feature that improves
availabilitv in Cisco Unified Communications Manager deployments.
1
As discussed before, IP phones that are located at remote locations can use an SRST gateway as
abackup for Cisco Unified Communications Manager in case of IP WAN failure The gateway-
can use its local dial plan to route calls that are destined for the IP phones in the main site over
the PSTN. But how should intersite calls be routed from the main site to the remote site while
the IP WAN is down?
The problem in this case is that Cisco Unified Communicalions Manager does not consider anv
other entries in its dial plan ifadialed number matches aconfigured but unregistered directory'
number. Therefore, if users al the main site dial internal extensions during the IP WAN outage
their calls will fail (or go to voice mail). To allow remote IP phones to be reached from the IP '
phones at the main site, you can configure CFUR for the remote-site phones CFUR should be
configured with the PSTN numbers that are used at the remote site so that internal calls for
remote IPphones are forwarded to the appropriate PSTN number. J
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Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
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52010 Cisco Systems. Inc.
J
Fallback for IP Phones: Fallback Mode
fhe figure illustrates the operation ofCisco Unified SRST when IP phones lose connectiv.h
tvith their primary- Cisco Unified Communications Manager.
Remote Site
Main Sile
Register
Cisco Unified
Communications
Manager
Remote
Gateway
When Cisco IP phones lose contact with Cisco Unified Communications Manager, they register
with the local Cisco Unified SRST router lo sustain the calI-processing capability that is
necessary to place and receive calls.
The Cisco Unified SRST gatewav automatically detects afailure, queries IP phones for
configuration, and automaticallv configures itself. The Cisco Unified SRST gateway uses
Simple Netuork-lnablcd Auto Provision (SNAP) technology to autoconfigure the branch
office router to prm ide call processing for Cisco IP phones that are registered with the router.
Cisco Unified Communications Manager Express in SRST mode can be used instead of
standard Cisco Unified SRST functionality. In this case. IP phones register with Cisco Unified
Communications Manager F.xpress when they lose the connection to their primary Cisco
Unified Communications Manager server. Cisco Unified Communications Manager F.xpress m
SRST mode provides more features than standard Cisco Unified SRSI.
Main Site I
Cisco Unified
Communications 1
Manager
Remote
Gateway
£
1
Fallback for IP phones is provided by the Cisco Unified SRST feature and improves the
availability of remote IP phones.
AWAN link connects IP phones at aremote site to the Cisco Communications Manager at a 1
central site, which is the call-processing device. Ifthe WAN link fails, Cisco Unified SRST
enables the gateway to provide call-processing services for IP phones. IP phones register with
the gateway (which is listed as abackup Cisco Unified Communications Manager server in the 1
server group configuration ofthe IP phones). The Cisco Unified SRST obtains the
configuration ofthe IP phones and can route calls between the IP phones or out to ihe PSTN.
The figure illustrates normal operation ofCisco Unified SRST while the connectivity between
IP phones and their primary server (Cisco Unified Communications Manager) is okay:
J
' «f!T?e 'Pph°nes ^ reSis,ered with Cisco lJn'ted Communications Manager over the IP
WAN.
J
• Cisco Unified Communications Manager manages call processing for IP phones.
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Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0 ©2010 Cisco Systems, Inc
MGCP Fallback: Fallback Mode
The figure illustrates operation of MGCP fallback in fallback mode-when the connectivity to
When the MGCP gatewav loses the connection to its call agent, it falls back to its detault call-
control application (POTS. H.323. or SIP). The gateway now uses alocal dial plan
configuration, such as dial peers, voice translation profiles, and so on. Ilence. .1 can operate
independent ofits MGCP call agent. Without MGCP fallback, the MGCP gateway would not
be able to process calls when the connection to its call agent is lost.
MGCP gateway fallback is afeature that improves the availability ofremote MGCP gateways.
AWAN link connects the MGCP gateway ataremote site to the Cisco Unified
Communications Manager at acentral site, which is the MGCP call agent. Ifthe WAN link
fails, the fallback feature keeps the gateway working as an H.323 or SIP gateway and re-homes
back to the MGCP call agent when the WAN link becomes active again.
The figure illustrates normal operation of MGCP fallback while the connectivity lo the call
agent (Cisco Unified Communications Manager) is okay:
• The MGCP gateway is registered with Cisco Unified Communications Manager over the IP
WAN.
• Cisco Unified Communications Manager is the call agent ofthe MGCP gateway that is
controlling its interfaces. The gateway does not have (or does not use) alocal dial plan
because all call-routing intelligence is at the call agent.
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Implementing Cisco Unified Communications Manager, Pari 2(CIPT2) v8.0 >2010 Cisco Systems, Inc.
PSTN Backup
The figure illustrates how calls can use the PSTN as abackup in case oflP WAN failure.
PSTN Backup
Main Site
Cisco Unified
Communications
Manager
1001-1099
1001-1099
In the example, calls to the remote site arc configured to use the IP WAN first and then use the
PS 1N as a backup option.
Availability Options
1 PSTN backup
1 MGCP fallback
1Fallback for IP phones:
- Cisco Unified SRST
- Cisco Unified Communications Manager Express
in Cisco Unified SRST mode
CFUR
• Mobility solutions: When users or devices roam between sites, they can lose features or
Fx^lnPrh l^ and^ati°nr,beCaUSe
Extension Mobility °f' Change
Cisco Dev.ce Mobility " th£irsuch
can solve actUal ^sicaI
issues. loc*ion.these
In addition Cisco
features allow integration ofcell phones and home office phones by enabling reachability
on any device via a single (office) number. «-^»auimy
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Implementing Cisco Unified Communications Manager, Part 2(CIPT2) vS.O
©2010 Cisco Systems, Inc.
Preventing Too Many Calls by CAC
This subtopic describes different methods of limiting the number ofconcurrent calls by CAC.
Preventing Too Many Calls by Cai
Admission Control
Cisco Unified Communications Manager allows the number of calls to be limited by these
CAC mechanisms:
• 1ocations' Cisco Unified Communications Manager location-based CAC is applicable to
calls between two entities thai arc configured in Cisco Unified Communications Manager.
These entities can be endpoints such as phones or devices that connect to other call-routing
domains such as trunks or gateways. However. CAC applies to the devices that are part o
the Cisco Unified Communications Manager cluster, even ifthey represent an external call
routing domain (in case of trunks). If ingress and egress device are in different locations,
the maximum bandwidth that is configured per location is checked at both ends, t alls
within a location arc not subject tothe bandwidth limit.
. Resource Reservation Protocol (RSVP)-cnabled locations: RSVP is aspecial way ,i
configure locations. When RSVP is configured to be used bclween apair ol locations, the
audio streams fious through two routers, so-called RSVP agents. The cal eg between the
t«o RSVP agents is subject to Cisco IOS RSVP CAC. Like with standard locations, ingress
and egress devices arc both part ofthe Cisco Unified Communications Manager cluster.
. Session Initiation Protocol (SIP) Preconditions: SIP Preconditions ««»lut«jn Hke
RSVP-enabled locations except that it is designed for SIP trunks on y With SH
Preconditions, calls through aSIP trunk flow through alocal Cisco IOSrouter at each end
ofthe SIP trunk splitting the call into three call legs-just like with RSVP-enabled
locations. However, in this case the call is not within acluster but bclween clusters.
. Gatekeepers- Gatekeepers arc used in the 11.323 world and provide address resolution and
CAC funSs HI 323 gatekeepers can be configured lo limit the number ofcalls between
H.323 zones.
Ifmulticast MOH from branch router fiash cannot be used (for instance, because the branch
router does not support the feature or does not have aCisco Unified SRST feature license) vou
can consider these alternatives:
• Using multicast MOH: When using multicast MOH over the IP WAN you can
significantly reduce the number of required MOH streams. Thus, less bandwidth is required
compared with multiple unicast MOH streams. The IP network, however, has to support
multicast routing for the path from the MOH server to the remote IP phones.
• Using G.729 for MOH to remote sites: Ifmulticast MOH is also not an option (for
instance, because multicast routing cannot be enabled in the network), you may still be able
to reduce the bandwidth that is consumed by MOH. When you change the codec that is
used for the MOH streams to G.729 and you potentially enable cRTP on the IP WAN each
individual MOH stream requires less bandwidth and hence reduces the load on the WAN
link. The bandwidth savings are identical to the bandwidth savings that you achieve when
"f'y/29 and cRTP for standard audio streams, which was discussed earlier To use
U729 for MOH streams, you have to put the MOH server and the remote IP phones into
different reg.ons. and you need to limit the audio codec between these two regions to 8
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©2010 Cisco Systems, Inc.
Multicast MOH from Branch Router Flash: Cisco IOS
Configuration Example
This subtopic shous the required Cisco IOS commands to enable multicast MOH from branch
router flash.
Cisco Unified
Com muni cation s
Manager
MOH
Configuration
DA 239 1 1 1
DP 16334
Ma" Hops
TTL 1
Main Site
_._ WAM.
dn 1
In the example, the name ofthe audio file on the branch router Hash is moh-lile.au. and the
configured multicast address and port number are 239.1.1.1 and 16384. respectively. Ihe
optional route command can be used to specify asource interface address lor the multicast
stream If no route option is specified, the multicast stream will be sourced from the configured
Cisco Unified SRST default address as specified by the ip source-address command under the
Cisco I'nificd SRST configuration (10.2.2.2 in this example). Note that you can stream only a
single audio file from flash and that you can use only asingle multicast address and port
number per router.
ACisco Unified SRST license is required regardless of whether the SRST functionality will
actually be used. The license is required because the configuration for streaming multicast
MOH from branch router flash is done in the SRST configuration mode and, even ,tSRSI
functionality will not be used, at least one IP phone (using the max-ephones command) and
one extension (using the max-dn command) must be configured.
6. The IP phone listens to the multicast MOH stream that was sent from the Cisco Unified
SRST gateway to IP address 239.1.1.1, port 16384, and plays the received MOH stream.
At no time do MOH packets cross the IP WAN.
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Multicast MOH from Branch Router Flash Example
The figure illustrates how multicast MOH from branch router Hash works.
When remote phone is put on hold Clko Unified Communicate Manager signals phone
to listen to DA 2391 1 1 DP 16384 andplay thereceived stream
Unified
Unified Communicalions Identical MOH
Communications Manager MOH Packets Packets Created
Manager MOH Dropped Here (TTL Here by SRST MOH
Configuralion Exceeded)
DA 239 1 1 1
DP 163B4
Max Hops
TTL 1
In the example the Cisco Unified Communications Manager MOH server is configured for
multicast MOH with adestination (multicast group) address of239.1.1.1. the destination port
16384. anda max-hops TTI.value of 1.
The Cisco Unilied SRST gateway that is located at the remote silc is configured with the same
destination IP address and port number as the Cisco Unified Communications Manager MOH
server.
• Configure an access control list (ACL) on the WAN interface: Configure an ACL onthe
WAN interface at the centralsite to disallow packetsthat are destined to the multicast
group address or addresses from being sent out the interface.
When you use multicast MOH from branch router flash. G.711 has to be enabled between the
Cisco Unified Communications Manager MOH serverandthe remote IP phones. This action is
necessary because the branchSRST MOH feature supportsonly G.711. Therefore, the stream
that is set upby Cisco Unified Communications Manager in thesignaling messages also has to
be G.711. Because the packets arenot sentacross the WAN, configuring the high-bandwidth
G.711 codec is not a problem as long as it is enabledonly for MOH. All other audio streams
(such as calls between phones) that are sent over the WAN should use the low-bandwidth
G.729 codec.
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Multicast MOH from Branch Router Flash
This subtopic explains how you can use multicast MOH from branch router fiash toreduce the
bandwidth that is required on the IP WAN.
Multicast MOH from branch router fiash is a feature for multisite deployments that use
centralized call processing.
Ihe feature works only with multicast MOI 1and isbased on MOH capabilities ofCisco
Unified SRST. The Cisco IOS SRSTgateway is configured for multicast MOH and
continuously sends a MOH stream, regardless ofitsSRST mode (standby orfallback mode).
In fact, neitherCisco Unified Communications Manager nor the remoteIP phones are aware
that the Cisco Unified SRST gateway isinvolved. To them, itappears asthough a multicast
MOH stream has been generated by the Cisco Unified Communications Manager MOH server
and has been receded by the remote IP phones.
Therefore, the remote IPphones are configured to use the centralized Cisco Unified
Communications Manager MOII server astheir MOH source, fhe Cisco Unified
Communications Manager MOH server isconfigured for multicast MOH (mandatory), and the
max-hops \alue in the MOI Iserver configuration is set to 1for the affected audio sources. The
max-hops parameter specifies the Time to Live (TTL) value that isused in the IP header ofthe
RTP packets, "fhe Cisco Uni lied Communications Manager MOI 1server and the Cisco IOS
SRST gatewav that is located at the remote site have to use the same multicast address and port
number for their streams. This way, MOH packets that are generated bythe Cisco Unified
Communications Manager MOH server atthe central site are dropped by the central-site router
because ITI. has been exceeded. As a consequence, the MOII packets donotcross the IP
WAN. The SRST gateway permanently generates amulticast MOH stream with an identical
multicast IP address and port number. The IP phone simply listens tothis stream as it appears
to becoming from the Cisco Unified Communications Manager MOH server.
Cisco Unified
Communications
Manager
-WAN-
Remote Site
In this example, ahardware conference bridge is deployed at the main site. The hardware
conference bndge is configured to support mixed conferences, in which members use various
codecs. Headquarters IP phones that join the conference can use G.711, while remote IP phones
canjoin the conference using a low-bandwidth codec.
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>2010 Cisco Systems, Inc.
Note Calls between IP phones at headquarters and remote IP phones do not require a
transcoder. They simply use the best allowed codec that is supported on both ends. G.729.
Atranscoder is invoked only when the two endpoints of acall cannot find acommon codec
that is permitted by region configuration. This principle is illustrated in this example. The
remote IP phones (which support G.711 and G.729) are not allowed to use GV11 over the
IP WAN and the headquarters voice-mail system and software conference bridge do not
support G729 Cisco Unified Communications Manager detects this problem that is based
on its region configurations, and the capability negotiation that is performed during call setup
signaling identifies the need for a transcoder. .
Bandwidth 8kb/s between BR and XCODER: This bandwidth ensures that the RTF
streams between remote IP phones and the transcoder. which are sent over the IP WAN. do
not use (J.711.
Bandwidth 64 kb/s between headquarters and XCODER: This bandwidth is required in
order for the G.711 -onh devices at headquarters lo be allowed to send G.711 to the
transcoder.
As a first step, jou need to implement the transcoding media resource. CiscoUnified
Communications Manager does not support software transcoding resources. Therefore, the only
option is to use a-hardware transcoding resource by first configuring the transcoder at the Cisco
IOS router and then adding the transcoder toCisco Unified Communications Manager.
The second step isto implement regions in a way that only G.729 ispermitted onthe IP WAN,
and the transcoder can be used ifrequired. To do so, you place all IPphones and G.711-only
devices, such as third-party voice-mail systems orsoftware conference bridges that are located
in the headquarters, in one region. You place remote IPphones in another region (called, for
example, branch, or BR). Thetranscoding resource is putinto a third region (called for
example. XCODER).
Now the maximum codec for calls within and between regions have tobe specified as follows:
• Bandwidth 64 kb/s within BR: This bandwidth allows local calls between remote IP
phones to use G.711.
• Bandwidth 64 kb/s within headquarters: This bandwidth allows local callswithin the
headquarters to use 0.711. These calls are not limited to calls between IP phones. They
also include calls to the G.711-only third-party voice-mail syslem orcalls that use the
G.711-only softwareconference bridge.
• Bandwidth 64 kb/s within XCODER: Because this region includes only the transcoder
media resource, this setting isnot relevant since there are no calls within this region.
• Bandwidth 8 kb/s between BR and headquarters: Thisbandwidth ensures that calls
between remote IP phones and headquarters devices (such as IP phones, software
conference bridge, and voice-mail system) do not use G.711 as the bandwidth for calls that
traverse the IP WAN is limited.
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Transcoders
The figure illustrates how you can use transcoders to reduce the bandwidth that isrequired on
the IP WAN.
Transcoders
In the example, a third-part; voice-mail system that supports only G.711 is deployed atthe
main site. One Cisco UnifiedCommunications Manager server is providing a software
conference bridge (which also supports G.711 only). Ifremote phones are configured lo use
G.729 over the IPWAN. they cannot join conferences oraccess the voice-mail system. To
allow these IP phones touse G.729 and toaccess the G.711-only services, you deploy a
hardware transcoder at the main site.
Remote IP phones now send G.729 voice streams to the transcoder over the IP WAN. The
transcoder changes the stream to G.711 and then passes iton to the conference bridge orvoice-
mail sNStem.
Each region in Cisco Unified Communications Manager isconfigured with the maximum audio
bandwidth requirements to be used per call:
• Within the configured region
• Toward a specific other region (manually configured)
• Toward all other regions (not manually configured)
Regions are assigned to device pools (one region per device pool), and adevice pool is
assigned to each device. Which codec isactually used depends on the capabilities ofthe two
devices that are involved in the call. The assigned codec is the one that is supported by both
devices and does not exceed the bandwidth requirements ofthe codec that is permitted in
region configuration. If devices cannot agree on a codec, a transcoder is invoked.
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Low-Bandwidth Codecs and RTP-Header Compression
The figure illustrates the effect of using RTP-header compression to conserve bandwidth on the
IP WAN.
82.4 EuHaiEIS2IuilE5Z3
kb/s 6 1 20 I 8 I 12 1 ieo
Cisco Unified
Communications
Manager
Remote Site
In the example, a \oice packet for a call that has default settings (G.711 codec and a 20-ms
packetization period) is being passed along aFrame Relay link. The frame has atotal size ot
206 Bcomprising 6Bof frame Relay header. 20 Bof IP header. 811 of UDP header. 12 Bof
RTP header, and 160 Bofdigitized voice, fhe packet rate is50 packets per second (p/s).
resulting ina bandwidth need of 82.4 kb/s.
When %ou use cRTP and change the codec toG.729. the required bandwidth changes as
follows: The frame now has atotal size of28 or30 Bper frame comprising 6bytes ofFrame
Relay header. 2or 4Bof cRTP header (depending on whether the UDP checksum is
preserved), and 20 Bof digitized, compressed voice. The packet rate is still 50 p/s (because the
packetization period was not changed), resulting in bandwidth needs of 11.2 or 12 kb/s.
Seven G.729 calls with cRTP enabled require less bandwidth than one G.711 call without cRl'P
(assuming that cRTP is used without preserving the UDP checksum).
Note While the audio codec configuration affects the end-to-end path, cRTP only affects WAN
links where cRTP isenabled RTP header compression isconfigured on a per link basis.
Deploying local music on hold (MOH) servers or using multicast MOH from branch
router flash: Deploying local MOH servers means thatCisco Unified Communications
Manager servers have to be present at each site. In centralized call-processing models in
which this requirement does not apply, itisrecommended that you use multicast MOH
from branch router flash. This approach eliminates the need ofstreaming MOH over the IP
WAN. Ifthis approach isnot an option, you should use multicast MOH instead ofunicast
MOH to reduce the number of MOH streams that have to traverse theIPWAN. Multicast
routing should be enabled in the network in order for multicast MOH function properly.
Limiting the number ofvoice calls using CAC: Use CAC to avoid oversubscription of
WAN bandwidth by too manyvoice calls.
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Solutions to Bandwidth Limitations
"fhis topic describes solutions to bandwidth limitations.
Cisco Unified
Communication a
Manager
As shown in the figure, if a local conference bridge is deployedat the remotesite, it keeps
voice streams off the IP WANfor conferences in which all members are physically located at
the remote site. You can implement the same solution for MTPs. MRGLs specify which
conference bridge (or MTP)should be used and by which IP phone.
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Disabled Annunciator
The figure shous howyou can conserve bandwidth on the IP WAN by sendingdisabling
annunciator streams to remote phones.
Disabled Annunciator
C'Sco Unified
Communicalions
Manager
If announcements should not be sent over the IP WAN, Media Resource Group Lists (MRGLs)
canbe used so thatremote phones do not have access lo the annunciator media resource.
Note Because not every call requires annunciator messages, and because the messages are
usually rathershort, the bandwidth that should be preserved by disabling the annunciator is
marginal.
QoS Advantages
Voice
(Highest) Voice is always served first
With QoS enabled, voice traffic is given absolute priority queuing ("PQ" in the figure) over all
other traffic. This approach prevents jitter, which is caused by variable queuing delays. It also
prevents lost voice packets, which are caused by tail drops that occur when buffers are
complete. To avoid the complete blocking ofother traffic, you should limit voice bandwidth.
The number ofvoice calls should also be limited by CAC so that there isnot more voice traffic
than there is bandwidth that has been reserved for it.
Finally, to ensure proper service for voice calls, you should configure QoS to guarantee a
certain bandwidth for signaling traffic. Otherwise, despite the fact that the quality ofactive
calls may be okay, calls cannot betorn down, and new calls cannot beestablished.
Note QoS is not discussed further in this course. For more information, refer to the Implementing
Cisco Voice Communications and QoS (CVOICE) course.
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Implementing Cisco Unified Communications Manager. Part 2(CIPT2) v8.0 ) 2010 Cisco Systems, Inc.
QoS
This topic describes how quality ofservice (QoS) can solve voice quality issues.
QoS Review
QoS refers to the capabilit\ ofanetwork lo provide belter service to selected network traffic.
The primarv goal of QoS is 10 provide better service, including dedicated bandwidth, controlled
jitter and latency {required bv some real-lime and interactive traffic), and improved loss
characteristics, by giving priority lo certain communication flows. It is also important to make
sure that providing priority for one or more flows docs not make other flows fail.
Fundamental. QoS enables vou to provide better service to certain flows. You can provide
better sen ice bv cither raising the priority ofaflow or limiting the priority ol another flow.
Some of QoS mechanisms are congestion management, congestion avoidance, and link
efficiencv
When you implement QoS. the implementation is split into three major steps:
• Traffic is identified (voice, signaling, data, and so on).
• Traffic is div ided into classes (real-time traffic, mission-critical traffic, less important
traffic, and soon).
• QoS policy is applied per class, specifying how to serve each class.
The figure illustrates amultisite deployment that incorporates the following solutions to
multisite deployment issues:
• Availability issues are solved by Cisco Unified Survivable Remote Site Telephony (Cisco
Unified SRST) and Media Gateway Control Protocol (MGCP) fallback.
• Quality and bandwidth issues are solved by quality ofservice (QoS), Call Admission
Control (CAC). Real-Time Transport Protocol (RTP)-header compression, and local media
resources.
• Dial plan solutions include access and site codes, as well as digit manipulation.
• Network Address Translation (NAT) and security issues are solved by the deployment ofa
Cisco Unified Border Element.
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Lesson 2
Identifying Multisite
Deplovment Solutions
Overview
Amultisite deplounent introduces several issues that do not apply lo single-site deployments.
When implementing Cisco Unified Communications Manager in amultisite environment, you
need to address these issues, fhis lesson provides information on how lo solve issues that arise
neei
in multisite deployments.
Objectives
Upon completing this lesson, you will be able to describe solutions for multisite deplovment
issues.
Summary
References
For additional information, refer to these resources:
• Cisco Systems. Inc. Cisco Unified Communications System 8.x SRND, April 2010.
hup://\vww.eisco.C(>m/en/US/docs/voice_ip_comm/cucni/srnd/8x/uc8x.html
• Cisco Systems. Inc. Cisco Unified Communications Manager Administration Guide
Release 8.0(1). February 2010.
http://uv\u .cisco.com/en/US/docs/voice.._ip_.comm/cucm/admm/8 0 l/ccmcfg/bccm-801
cm.himl
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Example: NAT Security Issues
fhe figure illustrates the private IP addresses ofCisco Unified Communications Manager
server and the IP phone that is being translated to public IP addresses.
Company A
Cisco Unified
Communications
Manager
In the example, both Companv Aand Company Buse IP network 10.0.0.0/8 internally. For the
companies to communicate over the Internet, the private addresses are translated to public IP
addresses. Companv Auses public IP network A. and Company Buses public IP network B.
All Cisco Unified Communications Manager servers and IP phones arc reachable from the
Internet and communicate with each other.
In these cases, or when connecting to apublic service such as an ITSP, you must configure
NAT tor Cisco Unified Communications Manager servers and IP phones. Once Cisco Unified
Communications Manager servers and IP phones are reachable with public IP addresses thev
will be subject to attacks from the outside world, which introduces potential securitv issues '
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Implementing Cisco Unrfied Communications Manager, Part 2(CIPT2) v8.0 ©2010 Cisco Systems, Inc.
The ideal solution for alarge deployment would allow an automatic recognition ofroutes.
Internal as well as external (for PSTN backup) numbers should be advertised and learned by
call-routing entities. Adynamic routing protocol for call-routing targets would address
scalabilit} issues in large deployments.
Call control discovcrv (CCD). a feature that is based on the Cisco Service Advertisement
Framework (SAF) provides such functionality. CCD and Cisco SAF are explained in more
detail in a later module of this course.