Ccvp plus module 2

2
Cisco IP Telephony Part 1
Faisal H. Khan
CCIE Voice Instructor

333
Course Flow Diagram
•Introduction to Cisco UCM
•Single-Site, On-Net Calling
•Single-Site, Off-Net Calling
•Implementation of Media Resources, Features and Applications
•Implementing user features

444
Cisco Unified Communications Architecture
– Core Call Processing capabilities on top of the Cisco IP network
infrastructure (IP Based PBX)
– Provide end point registration: IP Phone, Gateways, Voicemail
–Provide Dial Tone to IP Phone
–IP Phone Services for rich media capability
–Third party integration
–Video IP Telephony
–Contact Center

555
Cisco UCM Functions
–Call processing
–Signaling and device control
–Dial plan administration
–Phone feature administration
–Directory services
–Programming interface to
external applications
–Includes a backup-and-restore
tool (disaster recovery system)

666
Cisco UCM Signaling and Media Paths
Cisco UCM
IP
Phone A
Signaling
Protocol
(SCCP / SIP)
Media Exchange — (RTP)
Signaling Protocol
(SCCP / SIP)
 Cisco UCM performs call setup and maintenance tasks using a
Signaling Protocol (SCCP/SIP).
 Media exchange occurs directly between endpoints using RTP.
IP
Phone B

777
UCM Hardware/Cluster/OS
–Complete hardware and software solution (appliance model)
•Factory-installed and field-configured
•Can be installed on Cisco 7800 MCS server platform or on approved third-party
servers from IBM and HP
•No customer access to operating system
–Only GUI and CLI access to appliance system
–Third-party access via documented APIs only
–Supports clusters for redundancy and load sharing
•Provides database redundancy by sharing a common database
•Provides call-processing redundancy by Cisco UCM groups
•Cluster includes the following:
–One publisher
–Total maximum of 20 servers (―nodes‖) running various services, including
TFTP, media resources, conferencing, and call processing
» Maximum of eight nodes can be used for call processing (running the
Cisco UCM service)

888
Cisco Unified Communications Operating
System
–Appliance operating system (based on Red Hat Linux)
–Operating system updates provided by Cisco
(along with application updates)
–Unnecessary accounts and services disabled
–IDS as the database
–DHCP server
–Cisco Security Agent
–Cisco Unified Communications operating system is also used for
these other Cisco Unified Communications applications:
•Cisco Emergency Responder 2.0
•Cisco Unity Connection 2.0
•Cisco Unified Presence 6.0

999
Cisco Unified Communications Database
–IBM IDS database stores
•Static configuration data:
–Servers and enabled services within the cluster
–Devices (phones, gateways, and trunks)
–Users, dial plan, etc.
•Dynamic data utilized by user-facing features:
–Call Forward All, MWI
–Privacy, DND
–Hunt group login status, etc.
–Basically a single master database model
•R/W database access only for publisher (read-only for subscribers)
•Exception: Subscribers do allow R/W access for user-facing features
These features do not rely on the
availability of the publisher
because necessary data can be
written to subscribers.

101010
Database Access Control
–DB access between members of a cluster is protected
•By IP access control (dynamic firewall "iptables")
•By security password
–Special configuration procedure required to enable database access for subscribers
•At publisher, using Cisco UCM Administration, add subscriber to list of servers
before installation of subscriber
•During subscriber installation, enter same DB security password that was
configured during installation of publisher
Publisher Subscriber
Subscriber:
DB access
permitted
Other:
DB Access
Denied
Firewall

111111
Cisco UCM Licensing
•There are three types of licenses.
•Software license is required for using CUCM 6 software.
•Device license units required for devices (phones).
•Node licenses required for each call-processing Cisco
UCM server within the cluster.
•Licenses are required per cluster and provided by license
files.
•License file is bound to MAC address of publisher (running
the licensing service).
•Cisco Unified CM cluster continues to work if licensing
service is stopped (but no configuration changes allowed).

121212
Deployment Type
–Cisco UCM Deployment Options Cisco UCM Single-Site
Deployment
–Cisco UCM Multisite Deployment with Centralized Call
Processing
–Cisco UCM Multisite Deployment with Distributed Call
Processing
–Cisco UCM Multisite Deployment with Clustering Over the WAN
–Cisco UCM Call-Processing Redundancy

131313
Single-Site Deployment
–Cisco UCM servers, applications,
and DSP resources are at the same
physical location.
–IP WAN (if one) is used for data
traffic only; PSTN is used for all
external calls.
–Supports approximately 30,000 IP
phones per cluster. SIP/SCCP
Cisco
Unified CM
Cluster
PSTN

141414
Multisite WAN with Centralized
Call Processing
–Cisco UCM at central site; applications
and DSP resources centralized or
distributed.
–IP WAN carries voice traffic and call
control signaling.
–Supports approximately 30,000 IP
phones per cluster.
–Call admission control
(limit number of calls per site).
–SRST for remote branches.
–AAR used if WAN bandwidth is
exceeded.
SIP/SCCP
SIP/SCCP SIP/SCCP
PSTN
IP
WAN
Cisco
Unified CM
Cluster

151515
Multisite WAN with Distributed
Call Processing
–Cisco UCM and applications
are located at each site.
–IP WAN does not carry
intrasite call control
signaling.
–Gatekeepers can be used for
scalability.
–Transparent use of the PSTN
if the IP WAN is unavailable.
Gatekeeper
SIP/SCCP
SIP/SCCP SIP/SCCP
PSTN
IP
WAN
Cisco
Unified CM
Cluster
Cisco
Unified CM
Clusters
GK

161616
Clustering Over the IP WAN
–Applications and Cisco UCM of the same cluster distributed over
the IP WAN.
–IP WAN carries intracluster server communication and signaling.
–Limited number of sites.
Publisher /
TFTP
QoS Enabled BW
IP WAN
<40-ms Round-Trip Delay
SIP/SCCP
SIP/SCCP

171717
Cisco UCM Redundancy
–Maximum of eight call-processing servers in a cluster.
–Redundancy is provided by Cisco UCM groups.
•Prioritized list of call-processing servers (one or more).
•Multiple Cisco UCM groups can exist in the same cluster.
•Each call-processing server can be assigned to more than one
Cisco UCM group.
•Each device has a Cisco UCM group assigned determines the
primary and backup server to which it will register.

181818
Redundancy Design
 High availability (upgrade)
 Increased server count
 Simplified configuration
Primary
Secondary or
Backup
Publisher
and TFTP
Server (Not
Req. <2001)
Publisher
and TFTP
Server
Publisher
and TFTP
Server
7500 IP phones 15,000 IP phones 30,000 IP phones
Primary
1 to 7500
Backup
Backups
1 to
7500
1 to
7500
15001 to
22,500
7501 to
15,000
7501 to
15,000
22,501 to
30,000
Cisco 7845 Cisco 7845 Cisco 7845
Backups Backups

202020
Cisco UCM Installation and Upgrade Options
Option Description
Basic install
Install operating system and Cisco UCM application software
from bootable DVD.
Upgrade during
install
Basic install from bootable DVD; upgrade patches are installed
from FTP, SFTP, or local DVD.
Windows
upgrade
Upgrade from supported 4.x release. Existing database is
dumped to file server using the Data Migration Assistant tool.
Cisco Unified CM Release 6.x is installed from bootable DVD,
and data previously exported by DMA are imported into Cisco
Unified CM Release 6.x database.
5.x or higher
upgrade
Upgrade from 5.1(x) release or higher can be done from the
platform administration page using FTP or local DVD. Cisco
Unified CM software is updated; no installation from bootable
DVD is required.

212121
Important Configuration Information
Field Description
DHCP
Static or dynamic configuration of Server IP, hostname etc.
Options: Yes/No. If “No,” the hostname, IP address, IP mask,
and gateway have to be defined manually.
DNS Enabled
If DNS server exists in your network, enter Yes. When DNS is
not enabled, only IP addresses have to be used to reach all
network devices in your Cisco Unified Communications
network.
First Node If “Yes,” the first Cisco UCM node in the cluster is configured.
NTP
When enabled, this server will act as a NTP server and provide
time updates to the subsequent nodes in the cluster.
Security Password
Servers in the cluster use the security password to
communicate with one another. The password must contain at
least six alphanumeric characters.
SMTP
This field specifies the name of the SMTP host that is used for
outbound e-mail. You must fill in this field if you plan to use
electronic notification.

222222
Installation Procedures for Upgrade During
Installation
–Starting the installation.
•Boot the server with the installation DVD.
•Verify the checksum for the DVD.
•Choose to overwrite the hard disk.
–Platform Installation Wizard.
•Select Yes at the Apply Additional Releases window.
–Installation of operating system and application will start.
•When installation has completed, appliance will reboot.
–After reboot, choose Upgrade Retrieval Mechanism.
•Local: Specified path and file name.
•FTP/SFTP: Configure Network Settings and enter the
location and login information for the remote file server.

232323
Installation Procedures for Upgrade During
Installation (Cont.)
–Upgrade will start.
•When upgrade has completed, appliance will reboot.
–After reboot, at the Entering Pre-existing Configuration Information
dialog box, insert USB or disc if you have pre-existing configuration
information.
–Platform Installation Wizard.
•Select No at the Apply Additional Releases window.
•Select No at the Import Windows Data window
(if you have no existing Windows DMA data).
–Continue entering the Basic Install information if no USB or disc with
pre-existing configuration information has been inserted.
•Time zone, NIC, network settings, certificates, logins, passwords,
etc.
–Configuration scripts will run after the configuration information has been
collected, and network services will be restarted.

242424
Installation Procedures for
Windows Upgrade
–The Cisco Unified CallManager Release 4.x has to be backed up using
Cisco BARS.
–The Cisco Data Migration Assistant (DMA) is used to export the database
content to a file server.
–Installation of Cisco UCM Release 6.x.
•Server is booted with the installation DVD.
•The system hard disk needs to be overwritten.
–Platform Installation Wizard has to import Windows data.
–Installation of operating system and application will start.
–After completed installation, the Cisco DMA retrieval mechanism loads
the exported 4.x data file from these devices:
•A local path by file name.
•A FTP/SFTP server with given network settings, location, and login.

252525
Cisco Data Migration Assistant
• The Cisco Data Migration Assistant (DMA) is a tool for migrating
configuration information when upgrading from a Windows-based
Cisco UCM release to an appliance-based Cisco UCM release.
Cisco Unified
CallManager 4.2(3)
Publisher
Cisco Unified Communications
Manager Release 6.0(1) Publis
DMA
Cisco Unified
CM Release
6.0(1)
installation
imports file.
DMA exports
TAR file or
tape.
Network Share
Server
S/FTP
Appliance

262626
Cisco UCM Release 5.x and 6.x Upgrades
–Upgrades from Release 5.x or higher is done from the Cisco
Unified Operating System Administration page.
–Cisco UCM provides dual partitions.
•Holds two copies of the Cisco UCM software and database
(active and inactive partitions).
–Upgrade Process.
•Perform a backup using Disaster Recovery System (DRS).
•Start the installation of the new version (performed in the
background while current version is operating).
•After new version has been installed to inactive partition,
reboot, switching to new version.
•Cisco UCM will boot from partition where new version has
been installed.

272727
Dual Partitions
–Dual partitions each have UCM software and
database.
–Enables continued operation when you
upgrade software.
–Upgrade software installs on the inactive
partition.
–Activates the upgraded software by
―switching versions‖ during reboot.
–Current active partition becomes inactive and
retains current ―old‖ software until next
upgrade.
–If versions are switched before next upgrade,
you revert to previous version.
–System maintains two versions of software
(does not apply to Release 4.x upgrades).
Inactive
Partition
Active
Partition
5.1(1)
6.0(1)
5.1(1)
6.0(1)

28
Web interface for administration
Introducing VoIP

292929
Cisco UCM Administration and User Interface
Options

303030
Cisco UCM Administration CLI Main Page

31
Initial Configurations
CIPT 1

323232
Cisco UCM
Initial Configuration
Configure network settings NTP servers, DHCP services, remove DNS reliance
Verify network and Feature
services
Activate the necessary feature services and check
network services
Configure enterprise
parameters
Modify enterprise parameters as required
Configure service
parameters
Modify service parameters as required

333333
IP vs. DNS Considerations
• Cisco UCM Release 6.0 can use DNS names (default) or IP addresses
for system addressing.
Advantages of using IP addresses Advantages of using DNS
Does not require a DNS server
Simplifies management because of the
use of names instead of numbers
Prevents the IP telephony network from
failing if the IP phones lose connection to
the DNS server
Easier IP address changes because of
name-based IP paths
Decreases the amount of time required
when a device attempts to contact the
Cisco Unified CM server
Server to IP phone NAT possible
Simplifies troubleshooting

343434
Network and Features Services
Network Services Feature Services
Services required for the Cisco Unified
CM system to function; for example,
database and platform services.
Services that enable certain Cisco
Unified CM application features; for
example, TFTP, call processing, or
serviceability reports.
Automatically activated after Cisco
Unified CM installation. Cannot be
activated or deactivated.
Must be activated manually using
Unified CM Serviceability > Service
Activation.
Use Unified CM Serviceability >
Control Center > Network Services to
stop, start, or restart services.
Use Unified CM Serviceability >
Control Center > Feature Services to
stop, start, or restart services.

35
Managing user account
CIPT 1

363636
Two Types of User Accounts in Cisco UCM
End Users Application Users
Associated with an individual person Associated with an application
For personal use in interactive logins For non-interactive logins
Used for user features and individual
administrator logins
Used for application authorization
Included in user directory Not included in user directory
Can be provisioned and authenticated
using an external directory service
(LDAP)
Cannot use LDAP

373737
Data Associated with User Accounts
–Personal and organizational settings
•User ID, First Name, Middle Name and Last Name
•Manager User ID, Department
•Phone Number, Mail ID
–Password
–Cisco Unified CM configuration settings
•PIN and SIP digest credentials
•User privileges (user groups and roles)
•Associated PCs, controlled devices, and directory numbers
•Application and feature parameters (Extension Mobility
profile, Presence Group, Mobility, CAPF, etc.)

383838
User Privileges
–Privileges are assigned to application users and end users.
–Privileges include these accesses:
•Access to user web pages.
•Access to administration web pages.
–Access to specific administration functions.
•Access to APIs (CTI, SOAP, etc.)
–User privileges include these configuration elements:
•User groups (a list of application and end users).
•Roles (a collection of resources for an application).
–Each role refers to one application.
–Each application has one or more resources (static list).
–Per role, access privileges are configured per application resource.
•Roles are assigned to user groups.

393939
UserX
Admin
Group
Support
Group
Role1
Applicatio
n2
Resource1
Resource2
Resource3
Resource4
Applicatio
n1
Resource1
Resource2
Resource3
Applicatio
n1
Resource1
Resource2
Resource3
UserY
UserZ
UserC
Role2
Role3
read
(none)
read, update
read, update
read
(none)
read
(none)
read, update
read
User Privilege Component Interaction

404040
Roles and User Groups Example
–:SolutionObjective: Have administrators with full access and
administrators with read-only access to Cisco UCM
Administration
– Two user groups and two roles
Role Application PrivilegeUser Group
Standard
CCMADMIN
Administration
Cisco Unified
CM
Administration
Update
Standard
CCMADMIN
Read-Only
Read-Only
Standard CCM Super
Users
•User ―jsmith‖
•User ―mjane‖
Standard CCM
Read-Only
•User ―lukim‖
•User ―tedi‖
Resource
• Call Park
web pages
• AAR Group
web pages
• Cisco
Unified CM
Group web
pages
• DRF Show
Status Page
• …
Cisco Unified
CM
Administration

424242
LDAP
–Specialized database stores information about users
•Centralized storage of user information
•Available to all enterprise applications
–LDAPv3 – Lightweight Directory Access Protocol version 3
–Examples
•Microsoft Active Directory, Netscape, iPlanet, SunONE
–Cisco Unified CM supports two types of integration
•LDAP synchronization
•LDAP authentication
–When using LDAP, some user data are no longer controlled via
Cisco UCM Administration

434343
LDAP Integration Considerations
–Full synchronization.
•Microsoft Active Directory 2000
•Microsoft Active Directory 2003
–Incremental synchronization.
•Netscape Directory Server 4.x
•iPlanet Directory Server 5.1
•SunONE Directory Server 5.2
–All synchronization agreements must integrate with the same LDAP
family (Microsoft Active Directory or Netscape, iPlanet, and SunONE).
–Cisco Unified CM uses standard LDAPv3 to access data.
–One LDAP user attribute is chosen to map into the Cisco Unified CM
User ID field.

444444
Cisco Unified CM End-User Data Location
No LDAP
Integration
LDAP
Synchronization
LDAP
Authentication
Personal and organizational
settings:
User ID
First, Middle, and Last Name
Manager User ID and Department
Phone Number and Mail ID
Local
LDAP
(replicated
to local)
LDAP
(replicated
to local)
or
Local
Password Local Local LDAP
Cisco Unified CM Settings:
PIN and Digest Credentials
Groups and Roles
Associated PCs
Controlled Devices
Extension Mobility Profile and
CAPF Presence Group and Mobility
Local Local Local

454545
Cisco UCM BAT Characteristics
–Performs bulk transactions to the Cisco UCM database.
–Adds, updates, or deletes a large number of similar phones, users,
or ports at the same time.
–Exports data (phones, users, gateways, etc.).
•Exported files can be modified and re-imported.
–Integrated with the Cisco UCM Administration pages and
available by default (no plug-in required).
–Supports localization.
–Cisco Unified CM Autoregister Phone Tool (formerly TAPS) is
also available from the Bulk Administration menu but requires
additional products.

464646
Cisco Unified Communications Manager BAT
Configuration Process
• The Cisco Unified Communications Manager BAT
configuration procedure includes these steps:
–Step 1: Configure Cisco Unified CM BAT user template.
–Step 2: Create the CSV data input file.
–Step 3: Upload the CSV data input file.
–Step 4: Start Cisco Unified CM BAT job to add users.
–Step 5: Verify status of Cisco Unified CM BAT job.

474747
Configuration Methods and Tools
Method for Adding IP Phones Advantages Disadvantages
Autoregistration
 Devices
automatically
added
 Default Settings,
random DN
 Modifications needed
Unified CM BAT
 Bulk add  MAC addresses
required in BAT files
Unified CM Auto-Register
Phone Tool
 Very scalable
 MAC addresses
not required
 Cisco CRS required
 Complex configuration
Manual Configuration
 Simple  MAC addresses
required
 Time-consuming

484848
Endpoint Basic Configuration Elements
–Phone NTP Reference
–Date / Time Group
–Presences Group
–Device Pool
•Cisco Unified CM Group
•Regions
•Locations
–Security Profile
–Softkey Templates
–Phone Button Templates
–SIP Profile (SIP Phones Only)
–Common Phone Profile

494949
Phone NTP Reference
 Ensures that a
SIP phone gets
its date and
time from the
NTP server.
 If NTP servers
do not
respond, the
SIP phone uses
the date
header in the
200 OK
response.

505050
Date/Time Group Configuration
 Date/Time groups
define time zones
for devices
connected to Cisco
UCM.
 Date/time group is
assigned to device
pool.
 Device pool is
assigned to device.

515151
Device Pools
 Device pools
define sets of
common
characteristics
for devices.
 The device pool
structure
supports the
separation of
user and location
information.
 The device pool
contains only
device- and
location-related
information.

525252
Cisco Unified CM Group
 A Cisco Unified CM
Group specifies a
prioritized list of
up to three Cisco
UCMs.
 The first Cisco UCM
in the list serves
as the primary
Unified CM for that
group, and the
other members of
the group serve as
secondary and
tertiary (backup)
Unified CM.

535353
Regions
 Use regions to specify the bandwidth that is used for an audio or video call
within a region and between regions by codec type.
 The audio codec determines the type of compression and the maximum
amount of bandwidth that is used per audio call.

545454
Locations
 Use locations to implement
call admission control in a
centralized call-processing
deployment.
 Call admission control
enables you to regulate audio
quality and video availability
by limiting the amount of
bandwidth that is available
for audio and video calls.

555555
Phone Security Profile
 The Phone Security Profile
window includes security-
related settings such as
device security mode, CAPF
settings, digest
authentication settings (for
SIP phones only), and
encrypted configuration file
settings.
 You must apply a security
profile to each phone that is
configured in Cisco UCM
Administration.

565656
Device Settings
Device Settings contain default
settings, profiles, templates, and
common device configurations that
can be assigned to a device or device
pool.

575757
Device Defaults Configuration
Use device defaults to set the default
characteristics of each type of device that
registers with a Cisco UCM.

585858
Phone Button Template
Phone button templates specify how the phone
buttons of a Cisco IP phone should be used.
Options include lines, speed dials, and
functions such as callback, call pickup, etc.
Each Cisco IP phone has one phone button

595959
Softkey Template
Softkey template configuration allows the
administrator to configure softkey layouts which are
assigned to Cisco Unified IP phones.

606060
SIP Profile
A SIP profile comprises the set of SIP attributes that are
associated with SIP trunks and SIP endpoints. SIP profiles
include information such as name, description, timing,
retry, call pickup URI, and so on.

616161
Common Phone Profile
Common phone profiles include phone
configuration parameters and are assigned to
IP phones.

626262
Relationship Between Phone Configuration
Elements
NTP Reference Region
s
Locatio
ns
Common
Phone
Profile
SIP Profile
(SIP
Phones only)
Phone
Softkey
Templat
e
Date/Tim
e Group
Phone
Buttons
Templat
e
Device
Securit
y
Profile
Device
Pool

636363
Cisco UCM Endpoint Support
Cisco IP phone models displayed in italic are
end-of-sale.
Cisco Unified IP Phones
(SCCP and SIP)
Type A: 7940, 7960, 7905, 7912
Type B: 7906, 7911, 79[46][125], 797[015]
Cisco softphone Cisco IP Communicator
Other Cisco endpoints
(SCCP only)
7902, 7910, and 7931 (IP phones), 7920 and 7921
(WiFi phones), 7935 and 7936 (conference
stations), 7985 (desktop video phone)
Third-party endpoints
(various)
SCCP: Nokia dual-mode cell phone SCCP client,
Tandberg video endpoints, IP blue VTGO, etc.
SIP: various hard-and software phones
H.323: various hard- and software phones

646464
Unified CM Endpoint Telephony Feature
Support Dependencies
• Unified CM supports endpoints using SCCP, SIP, and H.323:
–Cisco proprietary SCCP:
•Only used by Cisco IP phones (few third-party endpoints exist)
•Rich set of telephony features, most features supported on all Cisco
IP phone models
–Standard SIP or H.323:
•Supported on all standard compliant third-party phones and few
Cisco IP phones
•Provide only basic telephony features
–Standard SIP with Unified CM extensions:
•Only used by Cisco IP phones
•Rich set of telephony features, but support depends heavily on Cisco
IP phone model

656565
Cisco SCCP IP Phone Startup Process
Unified CM Cisco TFTPDHCP
4
6
5
1 3
2
1. Cisco IP phone obtains power from the switch
2. Cisco IP phone loads locally stored image
3. Switch provides VLAN information to Cisco IP phone using
Cisco Discovery Protocol
4. Phone sends DHCP request; receives IP information and TFTP
server address

666666
Cisco SCCP IP Phone Startup Process (Cont.)
4
6
5
1 3
2
5. Cisco IP phone gets configuration from TFTP server
6. Cisco IP phone registers with Cisco UCM server
– Unified CM sends softkey template to SCCP phone using SCCP
messages.
Unified CM Cisco TFTPDHCP

676767
Boot Sequence Differences Between
Cisco SCCP and SIP Phones
• The boot sequences for SIP and SCCP are similar. The
first 4 steps remain the same. The main differences are :
–SEP<mac>.cnf.xml: The SIP phones get all of their configuration
from the configuration file. Therefore, the SEP<mac>.cnf.xml file
is much larger for SIP than for SCCP.
–Dialplan file (optional): The SIP phones can download and use
local dial plans.
–Softkey file: The SIP (Type-B only) phones download their softkey
sets in this XML file.

686868
H.323 Endpoints
–Cisco Unified IP Phone 7905 can be loaded with an H.323 firmware.
–From Cisco UCM perspective, they look like any other (third-party) H.323
endpoint.
–Other commonly used H.323 phones are Microsoft Windows NetMeeting
or H.323 video devices from vendors like Tandberg or Sony.
Cisco 7905 IP Phone
Third-Party H.323 Endpoints

696969
Features Not Supported for H.323 Endpoints
• H.323 phones only support a few features compared to Cisco IP
phones using SCCP or SIP. The features that are not supported
include but are not limited to:
–MAC address registration
–Phone buttons templates
–Softkey templates
–Telephony features and applications such as:
•IP phone services
•Cisco UCM Assistant
•Cisco Unified Video Advantage
•Call Pickup
•Barge
•Presence, etc.

707070
H.323 Phone Configuration Requirements
• H.323 endpoints typically require fewer configuration
steps on the Cisco UCM compared to other types of
endpoints. Configuration steps are as follows:
1. Configure the H.323 phone in Cisco UCM with IP address and
DN(s).
2. Configure the H.323 phone with the IP address of Cisco UCM and
specify the numbers that should be routed to Cisco UCM.

717171
Third-Party SIP Phone Support
–There are two categories of RFC 3261-compliant, third-party SIP
phones supported by Cisco Unified Communications Manager:
•Basic phones support one line and consume three license
units.
•Advanced support up to eight lines and video, and consume
six license units.
–Third-party SIP phones register with Cisco Unified
Communications Manager but are not recognized by a device ID
such as a MAC address. SIP Digest Authentication is used instead
to identify the endpoint that is trying to register.
–Configuration is performed on Cisco Unified Communications
Manager and on the phone itself.

727272
Third-Party SIP Phones
–Cisco Unified IP Phones 7940 and 7960 can be loaded with a
standard SIP software, which is different from using SIP with
Cisco Unified Communications Manager extensions on these
phones.
–From Cisco Unified Communications Manager perspective, these
phones look like any other (third-party) SIP endpoints.
–Many third-party SIP phones are available on the market.
Cisco 7960 IP Phone Third-
Party SIP Endpoints

737373
Features Not Supported for Third-Party SIP
Endpoints
• Third-party SIP phones only support a few features compared to
Cisco IP phones using SCCP or SIP. The features that are not
supported include but are not limited to the following:
–MAC address registration
–Phone button template
–Softkey templates
–Telephony features and applications such as:
•IP phone services
•Cisco Unified Communications Manager Assistant
•Cisco Unified Video Advantage
•Call Pickup
•Barge
•Presence

747474
SIP Digest Authentication
–Digest authentication provides authentication of SIP messages by
a username and a keyed MD5 hash.
–Digest authentication is based on a client/server model.
–Cisco Unified Communications Manager can challenge SIP
endpoints and trunks, but can only respond to challenges on SIP
trunks.
–Digest authentication is used to identify a third-party SIP device,
because no MAC address is provided in the registration message.
–Cisco Unified Communications Manager can be configured to
check the key (i.e. digest credentials) of a username used by a
third-party SIP device, or to ignore the key and only search for the
username.

757575
Third-Party SIP Phone Registration Process
Using Digest Authentication
directory number
= 2001
AuthID = ―sip‖
REGISTER 2001
username=―sip‖
Unified CM
Third-Party SIP Phone
End-user
config
―sip‖
Line
config
(2001)
Find
associated
device
Check directory
number and
accept
registration
Configuration
Database
Find end
user ―sip‖
Device
config

767676
Third-Party SIP Phone Configuration
Requirements
• The following steps have to be performed when
configuring third-party SIP endpoints:
1. Configure an end user in Cisco Unified Communications Manager.
2. Configure the third-party SIP phone and its directory numbers in
Cisco Unified Communications Manager.
3. Associate the third-party SIP phone with the end user.
4. Configure the third-party SIP phone with the IP address of Cisco
Unified Communications Manager (proxy address), end-user ID,
digest credentials (optional), and directory numbers.

77
Configuring Switch for Voice
CIPT 1

78
Enabling Single-Site On-Net Calling
Implementing MGCP Gateways in
Cisco UCM

797979
MGCP Gateways
–MGCP (defined under RFC 2705) is a master-slave protocol
–Allows a call control device (such as Unified CM) to take control of
a specific port on a gateway
–Provides centralized gateway administration and highly scalable
gateway solutions:
•Allows complete control of the dial plan from Unified CM
•Allows Unified CM per-port control of gateway connections to
PSTN, legacy PBX/VM systems, analog phones, etc.
–Allows use of plain-text commands between the Unified CM and
the gateway over UDP port 2427
–Gateway must be supported by Unified CM for MGCP
(use Cisco Software Advisor tool to verify compatibility)

808080
Endpoint Identifiers
MGCP
PSTN/
PBX
T1/E1 VWIC
2/1/1
FXS VWIC
2/1/1
AALN/S2/SU1/1@gw.voicebootc
amp.com
S1/SU1/DS1-
1@gw.voicebootcamp.com
AALN/S2/SU1/1@gw.voicebootc
amp.com
Endpoint
type
(analog
line)
Slot
2
Subunit
1
Port
1
S1/SU1/DS1-
1@gw.voicebootcamp.com
Slot 1 Subunit
1
Port
1
Endpoint type
(T1/E1 trunk)
Hostname
Hostname

818181
MGCP and SCCP Interaction
–Cisco IP phones use
SCCP to communicate
with Unified CM
–Unified CM uses
MGCP to control the
gateway
–Actual voice data is
through RTP directly
between the two
devices
Unified
CM Rel.
6.0
MGCP
PSTN
Gateway
SCCP
RTP/UDP

828282
Cisco UCM Configuration Server
135.1.1.1
135.1.1.101
T1/E1 VWIC
1/1/1
Unified CM
MGCP Gateway
TFTP
downloa
d
PSTN
Administrator configures
MGCP gateway in Unified CM
GW(config)#ccm-manager config server 135.1.1.1
GW(config)#ccm-manager config
Unified CM creates file
with MGCP configuration
for gateway
File is stored on Cisco
TFTP server
Gateway pulls
configuration file and
applies MGCP
configuration

838383
PRI Backhaul
–D-channel call-setup
signals need to be carried
in their raw form back to
the Unified CM to be
processed
–Gateway terminates data
link layer and passes the
rest of signals (Q.931 and
above) to Unified CM via
TCP port 2428
–D-channel will be down
unless it can
communicate with
Unified CM
PRI
Backhaul
T1 PRI
ISDN
Call Ctrl
Q.931
TCP Q.921TCP
Q.931
Q.921
CUCM Gateway PSTN CO

84
Enabling Single-Site On-Net Calling

858585
Endpoint Addressing Characteristics
–Reachability of internal destinations is provided by assigning directory
numbers
–Directory numbers are assigned to endpoints (phones, fax machines, etc.)
and applications (voice mail systems, auto attendant, etc.)
–The number of extensions required generally determines the length of
directory number digits
–DID numbers for inbound PSTN calls are mapped to internal directory
numbers
3001 3002 3003 30053004
Cisco
Unified CM Cisco
Unity

868686
Endpoint Dialing
–On-Net Dialing: Calls that
originate and terminate on the
same telephony network (e.g.,
internal IP phone to IP phone
calls within the same cluster)
–Off-Net Dialing: Calls that
originate from a telephony
network and terminate on a
different telephony network
(e.g., IP phone to PSTN calls)
–Abbreviated Dialing: Use of
internal number to reach a
PSTN phone. Unified CM
maps the abbreviated number
to full PSTN number
2001 20032002 2004
PSTN
416-555-
4001
dials 4001

878787
PSTN
Endpoint Dialing Example
3001 3002
HQ
Site 1
dials 3001
4001 4002
Site 2
dials 4001
416-555-4001
On-net
Abbreviated
555-2001
dials 9
5552001
Off-net
2001 2002 2003
IP WAN

888888
Uniform On-Net Dial Plan Example
Range Use DID Ranges Non-DID Ranges
0XXX
Excluded: 0 is used as Off-
Net access code
1XXX Site A extensions 418 555 1 XXX N/A
2XXX Site B extensions 919 555 2XXX N/A
3XXX Site C extensions 415 555 30XX 3[1-9]XX
4[0-4]XX Site D extensions 613 555 4[0-4]XX N/A
4[5-9]XX Site E extensions 450 555 4[5-9]XX N/A
5XXX Site A extensions 418 555 5XXX N/A
6XXX Site F extensions 514 555 6[0-8]XX 69XX
7XXX Future
8XXX Future
9XXX
Excluded: 9 is used as Off-
Net access code

898989
Call Routing Types
Routing Type Routing Component and Characteristics
Intrasite
 Calls within a single site (on-net)
 Uses assigned directory numbers to route calls internally
 Directory numbers usually have uniform length
Intersite
Calls between sites:
 On-net: Uses internal directory numbers
 Off-net: Uses route patterns to send calls to other site
through PSTN gateway; if abbreviated dialing is used,
internal number has to be translated to PSTN number
first
PSTN
Calls to PSTN (off-net)
 Uses route patterns to send calls to PSTN destinations

909090
Call Routing Table Entries (Call Routing
Targets)
Routing
Component
Description
Directory
Numbers
Numbers assigned to all endpoints and applications; used for
internal routing within a cluster
Translation
Pattern
Used to translate a dialed number and then look up the
translated number in the call routing table again
Route Pattern
Used to route calls to off-net destinations (via a gateway) or to
other Unified CM clusters (via a trunk)
Hunt Pilot
Used to route calls to hunt group members based on a
distribution algorithm (longest-idle, circular, etc)
Call Park
Numbers
Allows placing a call on hold to a number and retrieving back
the call from other phone by dialing the number
Meet-Me
Numbers
Allows a conference call initiator to set up a conference call
and attendees to join the conference by dialing the conference
number

919191
Sources of Call Routing Requests (Entities
Requiring Call Routing Table Lookup)
Routing
Component
Description
IP Phones
A number dialed by an IP phone is looked up in the routing
table.
Trunks
A call request received through a trunk is looked up in the
routing table.
Gateways
A call request received from a gateway is looked up in the call
routing table.
Translation
Patterns
After a translation pattern was best matched (as a target of a
call routing table lookup), the transformed number is looked up
again in the call routing table. The entity that generates this
lookup is the translation pattern.
Voice Mail Ports
A voice mail system can be configured to allow calling other
extensions or PSTN numbers (e.g., the mobile phone of an
employee). In these cases, the call routing request is received
from the voice mail port of Unified CM.

929292
Route Pattern: Commonly Used Wildcards
Wildcard Description
x Single digit (0–9, *, #)
@ North American Numbering Plan
! One or more digits (0–9)
[x-y] Generic range notation
[^x-y] Exclusion range notation
. Terminates access code
# Terminates interdigit timeout
<wildcard>?
Matches zero or more occurrences of any digit that matches the
previous wildcard
<wildcard>+
Matches one or more occurrences of any digit that matches the
previous wildcard

939393
Route Pattern Examples
Pattern Result
1234 Matches 1234
1*1x Matches numbers from 1*10 to 1*19
12xx Matches numbers from 1200 to 1299
13[25-8]6 Matches 1326, 1356, 1366, 1376, 1386
13[^3-9]6 Matches 1306, 1316, 1326, 13*6, 13#6
13!#
Matches any number that begins with 13, is followed by one or
more digits, and ends with #; 135# and 13579# are example
matches

949494
Digit-by-Digit Analysis
Route Patterns
1001
2001
Dialed Digits
<none> List Potential
Matches
1 List Potential
Matches
0 List Potential
Matches
0 List Potential
Matches
1 List Current
Match
Call Setup
1XXX
10XX

959595
Digit Collection Example
1111
121X
1[23]XX
131
13!
13[0-4]X
User dial string: Match!
Does not match
Does not match
Does not match
Does not match
Does not match
No other patterns could
match; extend call.
Cisco Unified CM actions:
1111

969696
Cisco UCM Addressing Method
Device Signaling Protocol Addressing Method
IP Phone
SCCP Digit-by-digit
SIP
En-bloc
KPML
SIP dial rules
Gateway MGCP/SIP/H.323
En-bloc
Trunk SIP, H.323
En-bloc

979797
User Input on SCCP Phones
–SCCP Phones report every input event (off-hook, on-hook, each
digit dialed, etc.) to Unified CM immediately.
–Unified CM analyzes phone input digit-by-digit against configured
dial plan and responds with feedback (dial tones, ring back,
reorder tone, etc.).
–No dial plan information at the IP phone.
SCCP message sent
with each user action
Dial Plan
(digit analysis)
Off-hook, digit 1, digit 0, digit0, digit 0
Dial tone on/off, screen update. etc.
Any phone model
running SCCP.
Signaling
Dialing actions:
1 0 0 0

989898
User Input on SIP Phones
–Type A SIP phones
•Cisco Unified IP phones 7905, 7912, 7940, and 7960
•Do not support KPML
–Type B SIP phones
•Cisco Unified IP phones 7911, 7941, 7961, 7970, and 7971
•Support KPML
–SIP dial rules can be configured on both phone types

999999
User Input on Type A SIP Phones – No SIP Dial
Rules Configured on the Phone
–Phone accumulates all user input events until # or Dial softkey is
pressed (similar to with cell phones)
–Phone will send SIP INVITE message with complete dialed digits
(en-bloc)
–Unified CM analyzes the full dialed digits against configured dial
plan
SIP INVITE message
sent when user presses
the Dial key
Dial Plan
(digit analysis)
―call for 2001‖
Call in progress, call connected, call denied, etc.
Existing SIP
phone
such as 7940,
7960
Signaling
Dialing actions:
2 0 0 1 Dial

100100100
User Input on Type A SIP Phones – SIP Dial
–SIP dial rules enable phone to recognize patterns dialed by users
–If pattern matches, SIP INVITE will be sent immediately without
requiring user to press # or Dial softkey
–The phone below is configured to immediately recognize all
four-digit patterns beginning with 1 (timeout value of 0 for 1…)
SIP INVITE message
sent when pattern
is recognized
Dial Plan
(digit analysis)
―call for 2001‖
Existing SIP
phone
such as 7940,
7960
Signaling
Dialing actions:
20 0 1 Dial
Pattern 1…
Timeout 0

101101101
User Input on Type B SIP Phones – No SIP Dial
–Based on KPML to report user key presses, every user key press
triggers a SIP NOTIFY message to Unified CM
–Very similar behavior to phones running SCCP
–No Dial softkey to indicate the end of user input
KPML events reported
in SIP NOTIFY messages
Dial Plan
(digit analysis)
Off-hook, digit 1, digit 0, digit 0 , digit 0,
SIP enhanced
phone
such as 7971
Signaling
Dialing actions:
2 0 0 1 Dial

102102102
User Input on Type B SIP Phones – SIP Dial
–Combination of KPML and SIP dial rules will be used
–Dial rules are processed first
•Once dial rule is matched, appropriate digits are sent en-bloc
•If additional digits are required, KPML is used
•Additional digits are sent one-by-one using KPML
SIP INVITE message
sent when pattern
is recognized
Dial Plan
(digit analysis)
―call for 2001‖
Signaling
Dialing actions:
2001 Dial
Pattern 2…
Timeout 0
SIP enhanced
phone
such as 7971

103103103
Path Selection
–Path selection is an essential dial plan element.
–After call routing decision is done, where should the call be
sent to?
–Chooses the best path:
•Which device to use (gateways, trunks, etc.)?
•Backup path available if first choice not available?

104104104
Routers/Gateways
Path Selection Example
–For off-net calls, a route pattern must be configured on Unified CM
–In above example, to reach 416-526-4000, use:
1. IP WAN through an ICT as priority path.
2. If WAN not available, try the second path through PSTN.
416-526-4000
San Jose
PSTN
IP WAN
Gatekeeper
1
2
User dials 9-
1-416-526-4000
1001
GK

105105105
Route
Pattern
Route
List
Route
Group
Second
Choice
Route
Group
First Choice Second
Choice
ConfigurationOrder
 Matches dialed number for external calls
 Performs digit manipulation (optional)
 Points to a route list for routing
 First level of path selection
 Performs digit manipulation
 Points to prioritized route group(s)
 Second level of path selection
 Points to the actual device(s)
PSTNIP WAN
First
Choice
Route pattern:
Route list:
Route group:
 Gateways (H.323, MGCP)
 Trunks (SIP, H.323)
Devices:
Path Selection Configuration Elements in Cisco
UCM
GK

106106106
Route List
User dials
914165265000
PSTNRoute Group
GW 1
GW 2
Route Group Configuration
• A route group is a list of devices that share
the same requirements for digit manipulation
(e.g., multiple PSTN gateways).
Gateway pulls
configuration file
and applies MGCP
configuration
Circular (round-
robin) or top down
(priority-based)
distribution
algorithm can be
configured
Route Pattern
9.14165265XXX

107107107
Trunk
GW-B
GW-B
Route List
First
Choice
Second
Choice
Route Group
IP WAN
Route Group
PSTN
Route List Configuration
PSTN
IP
A route list is a prioritized list
of route groups.
User dials
914165264000
Route Pattern
9.14165264XXX

108108108
The @ Wildcard
–Macro function that expands into a series of route patterns
–Represents the entire national numbering plan for a certain
country
–Example, configuring a 9.@ route pattern adds 166 individual
NANP route patterns to Unified CM database
–It is possible to modify and use @ for other country numbering
plan
–Can be used with route filters to block certain components of the
number

109109109
Route Filters
–Used only with @ route pattern to block certain patterns (e.g., block
all 1-900 calls, etc.) defined by clauses
–Not recommended for large deployments; use explicit route patterns
rather than @ wildcard
–Match clauses are based on tag operators and values
–Example, Match all NANP dialed numbers that include area code 416
(e.g., 9.14165551234)
•Route pattern: 9.@
•Route filter: IF AREA-CODE = 416
–Example: Match all NANP dialed numbers that include the selection
of a long-distance carrier (e.g., 9.101044414165551234)
•Route pattern: 9.@
•Route filter: IF TRANSIT-NETWORK EXISTS

110110110
The ! Wildcard
–Stands for one or more digits
–Used for variable-length route patterns (e.g., some international calls)
–Subject to T302 timer (post-dial delay)
•15 seconds by default
•T302 timer can be configured (typically reduced):
–Service Parameter > Call Manager > Clusterwide parameters
(Device – General)
–Users can indicate end of dialing by pressing #
•Requires an identical route pattern with # wildcard at the end
•Different behavior compared to Cisco IOS dial peers
•In Unified CM, # is seen as part of dialed string (therefore, if
used, it does not match route pattern without #)

111111111
Urgent Priority
–Configured under Route Pattern configuration
–Used to force immediate routing as soon as match is detected –
even if other, longer route patterns are potential matches
–Used with emergency number route patterns
–Effectively excludes the urgent pattern from a longer route pattern
range
–Translation patterns always have urgent priority

112112112
Blocked Patterns
–A route pattern can be configured for either ―Allow‖ or ―Block‖.
–Block patterns will prevent calls to the pattern cluster-wide.
–The same can be configured on translation patterns.

113113113
Call Classification
–Classify a call as on-net or off-net
–Configured on route patterns for outgoing calls and devices
(trunks and gateways) for incoming calls
–―Allow device override‖ setting uses the classification of the used
device on outgoing calls (rather than route pattern classification)
–Used by several features:
•Blocking off-net to off-net transfers (toll-fraud prevention)
•Drop conference when no on-net party remains
•Call forward external versus call forward internal

115115115
Digit Manipulation
Cisco IP Phones
CCM1-1
SIP 3rd party
IP Phone
T1/E1
Off-Net
Calls
Local
Gateways
PSTN
1002
416-555-1111
DID:
706-
555-
1001
to
1003
How to
Manipulate
Calling and
Called Number?
 Expand calling
directory
number to fully
qualified PSTN
number
 Strip access
code 9 dialed
internally for
PSTN access
On-Net Off-Net
Calling 1002 706-555-1002
Called
9.1416-555-
1111
1416-555-1111
CCM2-1

116116116
Digit Manipulation Requirements
Requirement Call Type How
Expand calling-party
directory number to full
E.164 PSTN number
Internal to PSTN
Use calling party’s external
phone number mask or calling
party transformation in route
pattern or route list
Strip PSTN access code “9” Internal to PSTN
Use Digit Stripping in Route
Pattern or Route List
Expand abbreviated number
(e.g., “0” for operator)
Internal to Internal
Use Called Party Transformation
in Translation Pattern
Convert E.164 PSTN called-
party directory number to
internal number
PSTN to Internal
in Translation Pattern, or use
Significant Digits
Overlapping endpoint
directory number
Internal to Internal
PSTN to Internal
in Translation Pattern

117117117
PSTN1005
303-555-
6007
416-555-
30xx
GW 416-555-3005 is
calling
Dials: 9-1-303-555-
6007
Digit Manipulation Flow Example (Outgoing
Call to PSTN)
Step Description
1 Extension 1005 dials 9-1-303-555-6007
2
Dialed number matches 9.! Route pattern configured with the following:
– Called party transformations > Discard digits: PreDot
– Calling party transformations: 41655530XX
– Route to GW
3
Unified CM strips off (discards) digit 9 from the dialed number and sends
13035556007 to PSTN via the GW after modifying the calling party number
from 1005 to 4165553005
4 PSTN phone 3035556007 rings and sees 4165553005 as the calling number

118118118
Digit Manipulation Flow Example (Incoming
Call from PSTN)
Step Description
1 PSTN phone dials 1-416-555-3010, PSTN switch routes the call to GW/Unified CM
2
Incoming call dialed number matches 41655530XX translation pattern configured with the
following:
– Called Party transformation > Called Party Transform Mask: 10XX
– (Optional) Calling Party transformation > Prefix Digit: 91
3
– Unified CM translates 4165553010 to 1010
– Unified CM looks up 1010 and finds a registered phone with that directory number
4
Unified CM presents the call to extension 1010. It will (optionally, see Step 2) prefix the
calling number with 91 to make it easier for the internal user to call back the PSTN caller
from IP phone Directory button (no need to manually add 91)
PSTN
1010
416-555-
30xx
GW
Dials: 1-416-
555-3010
303-555-
6008

119119119
Digit Manipulation Configuration Elements
Digit Manipulation Element Characteristics
External Phone Number Mask
Designates the fully qualified E.164 address for the
user extension
– Part of Calling/Called Transformation settings.
Digit Prefix and Stripping
Prefix or strip dialed digits from a route or
translation pattern for outbound calls
Transformation Masks
Manipulate the dialed digits or calling party number
Translation Pattern
When dialed digits match the translation pattern,
Unified CM performs the translation first and then
routes the call again.
Make use of the Calling/Called Transformation
settings for digit manipulation.
Significant Digits
Strip off digits received by Unified CM for incoming
calls from a PSTN gateway or from a trunk.

120120120
External Phone Number Masks
–Designates the fully qualified E.164 address for the user extension
–Used to format caller ID information for external (outbound) calls
that are made from the internal devices
–Configured under Line Configuration settings, but enabled as part
of Calling Party Transformations settings.

121121121
Configuring External Phone Number Mask
–Go to Device > Phone > Find
and select the corresponding
phone
–Under Association Information,
click the corresponding Line
–Scroll down to Line x on Device
configuration (see picture)
–Type full E.164 PSTN number
in the External Phone Number
Mask field
–In the Route Patterns that point
to PSTN (e.g. 9.! or 9.@), scroll
to Calling Party
Transformations
–Check the Use Calling Party's
External Phone Number Mask

122122122
Digit Prefix
–Prepend digits to the pattern
–Valid entries include the digits 0 through 9, *, and #
–Part of Calling/Called Transformations settings

123123123
Digit Stripping
–Used to strip digits from a pattern
–Part of Called Party Transformations settings (Discard Digits
field)
–A discard digits instruction (DDI) removes a portion of the dialed
digit string before passing the number on
–If no @ sign (numbering plan) is used in route pattern, only the
following DDIs are supported:
•PreDot
•NoDigits
DDI

124124124
Discard Digits Instructions (DDIs)
For example, If the pattern is 9.5@
Instructions Discarded Digits Used for
PreDot 95 1 214 555 1212
Removes access code digit(s)
delimited by . sign
PreAt 95 1 214 555 1212
Removes all digits that are in front
of a valid numbering plan pattern
11D/10D@7D 95 1 214 555 1212
Removes PreDot/PreAt digits and
local or long-distance area code
11D@10D 95 1 214 555 1212
Removes long distance area code
identifier (1)
IntlTollBypass 95 011 33 1234 #
Removes international access
(011) and following country code
10-10-Dialing 95 1010321 1 214 555 1212
Removes carrier access (1010)
and following carrier ID code
Trailing-# 95 1010321 011 33 1234 #
Removes of dialed # sign (to
terminate dialing without timeout)

125125125
Using PreDot DDIs
PBX
Unified CM Match: 9.8XXX
Discard: PreDot
Called Party: 8123
User Dials: 98123

126126126
Using Compound DDIs
• Use DDIs to remove
carrier selection from
dialed number. Carrier
selection consists of:
– Carrier Access Code: 1010
– Carrier Identification Code:
3 digits
Match: 9.@
Discard:
PreDot 10-10-Dialing
User Dials:
9-1010-288-1-214-555-1212
Called Party:
12145551212
Unified CM
PSTN

127127127
Transformation Settings
–Calling Party Transformations control the adaptation of calling
party numbers from enterprise format to PSTN format
–Called Party Transformations manipulate the dialed digits,
Number Type, and Numbering Plan.

128128128
Calling Party Transformation Order
41685XX000
1.Apply the external
phone number mask
2.Apply the calling party
transformation mask
3.Apply prefix digits
35062
21471XXXXX
41685XX000
2147135062
4168535000
Directory Number
External Phone
Number Mask
Calling-Party
Transformation
Mask
Caller ID
√

129129129
Called Party Transformation Order
1. Apply discard digits
2. Apply the called-party
transformation mask
3. Apply prefix digits
9 1010321 18085551221
10-10-Dialing
XXXXXXXXXX
9 18085551221
8085551221
Dialed
Number
Discard Digits
Called-Party
Transformation
Mask
Prefix Digits
Called Number 88085551221
8

131131131
Calling Privileges
• Calling privileges (also called class of service) define the
entries of a call routing table that can be accessed by an
endpoint performing a call routing request.
–Used to control telephony charges
•Block costly service numbers
•Restrict international calls
–Used for special applications including:
•Route calls with the same number differently per user
(different gateway per site for PSTN calls)
•Route calls to the same number differently per time of day

132132132
Call Privileges Requirement Example
Calling Privilege Class
(Class of Service)
Allowed Destinations
Internal
 Internal
 Emergency
Local
 Internal
 Emergency
 Local PSTN
Long Distance
 Internal
 Emergency
 Local PSTN
 Long Distance PSTN
International
 Internal
 Emergency
 Local PSTN
 Long Distance PSTN
 International PSTN

133133133
Call Privileges Configuration Elements
Call Privileges Element Characteristics
Partitions
Group of numbers (directory numbers, route patterns,
translation patterns, etc.) with similar reachability
characteristics
Calling Search Spaces
(CSSs)
Defines which partitions are accessible to a particular
device
Time Schedules and
Time Periods
Used to allow certain partitions to be reachable only during
a certain time of the day
Client Matter Codes
(CMC)
Used to track calls to certain numbers
A user must enter a Client Matter Code to track calls to
certain clients
Forced Authorization
Codes (FAC)
Restrict outgoing calls to certain numbers
A user must enter an authorization code to reach the
number

134134134
Partitions and Calling Search Spaces
–A partition is a group of numbers with same reachability.
•Any dialable patterns can be part of a partition (directory
numbers, route patterns, translation patterns, voice-mail ports,
Meet-Me conference numbers, etc.).
–Calling search space is a list of partitions and includes the
partitions that are accessible by this CSS.
•A device can call only those numbers located in the partitions
that are part of its calling search space.
•Assigned to any entity that can generate a call routing request,
including phones, phone lines, gateways, and applications.

135135135
Phones Have a Device CSS and
Line CSS
• IP phones can have a
CSS configured at each
line and at the device.
–CSS of the line from
which the call is placed
is considered first
–Device CSS is then
added
–Effective CSS consists
of:
1. Line CSS
2. Device CSS
Partition D1
Partition D2
Partition D3
Device CSS
Partition L1
Partition L2
Partition L3
Line CSS
Partition L1
Partition L2
Partition L3
Resulting CSS
Partition D1
Partition D2
Partition D3
Line
Device

136136136
Time-of-Day Routing Overview
–Time and date information can be applied to partitions.
–CSSs that include such a partition only have access to the partition
if the current date and time match the time and date information
applied to the partition.
–Allows different routing based on time
•Identical route pattern is put into multiple partitions.
•At least one partition has time information applied.
•If this partition is listed first in CSSs, it will take precedence
over other partition during the time applied to the partition.
•If time does not match, second partition of CSS is used
(first one is ignored due to invalid time).

137137137
Time Periods and Time Schedules
• Time period
–Time range defined by start
and end time
–Repetition interval—Days
of the week or specified
calendar date
–Associated with time
schedules
• Time schedule
–Group of time periods
–Assigned to partitions
–Determines the partitions
that calling devices search
when they are attempting to
complete a call during a
particular time of day
Partition
weekdayhrs_TP 0800–1700 M – F
weekendhrs_TP 0800–1700 Sat –
Sun
newyears_TP 0000–2400 January
1
noofficehours_TP
Sat – Sun
weekdayhrs_TPRegEmployees_TS
CiscoAustin_PT RegEmployees_TS
Start–End Repetiti
on
Time
Periods
Time Schedule
Time Schedule
Time
Periods

138138138
Time-of-Day Routing Configuration Procedure
1. Create time periods.
2. Create time schedules.
3. Assign time schedules to partitions.

139139139
Client Matter Codes and
Forced Authorization Codes
–CMC: Forces the user to enter any
configured CMC
•Allows for billing and tracking of calls
made per client
–FAC: Forces the user to enter a
configured authorization code with a
high-enough authorization level
•Prevents unauthorized user from
making toll calls
•Can be combined with time-of-day
routing (e.g., international calls outside
business hours require FAC)
–Both generate Call Detail Records

140140140
CMC Call: Successful Call
1. Dial number that goes
to CMC-enabled route
pattern
2. Unified CM tells phone
to play tone to prompt
for CMC
3. User enters valid code
number
4. Call extended
5. Generate CDR for
billing
CMC:
1234
1244
3489
User A Voice GW

141141141
FAC Call: Successful Call
User A
1. Dial number that goes to
an FAC-enabled route
pattern
2. Unified CM tells phone
to play tone
3. User enters
authorization code
4. Code is known and
authorization level is
not lower than required
level configured at
route pattern
5. Call extended
6. Generate CDR
Voice
FAC:
1234: Level
1
1244: Level
2
1888: Level
7

142
Call Forwarding, Shared Lines, and Call
Pickup

143143143
Call Forwarding
–CFA, CFNA, and CFB are configured under directory number
settings.
–CFA is configurable by end user from phone or user web page.
–CFNA and CFB are configurable by end user from user web page.
–If CFA is configured, the call will be forwarded immediately to the
configured number. The forwarding IP phone will not ring.
Voice Mail
2000
2001
User dials
2000
91551234
CFA
(All)
CFB (Busy)
CFNA
(No Answer)

144144144
Shared Lines
–Same directory number configured on multiple phones.
–All phones will ring at the same time if directory number is called.
–A user will pick up the call from one of the phones. All phones stop
ringing when the call is answered.
All 3 phones will ring
2000
2000
2000
2
User dials
2000
1

145145145
Call Pickup/Group Call Pickup
• Multiple lines can be grouped together into a pickup
group
–Each pickup group is identified by a unique pickup group
number.
–Each phone line can be a member of one pickup group.
• Call Pickup
–Allows a user to answer a call that is ringing on a phone in the
same pickup group as the phone of the user.
• Group Call Pickup
–Allows a user to answer a call ringing on any phone that is in a
different pickup group than the phone of the user.
–Requires the user to enter the pickup group number.

146146146
Line Group 1
2001 1001
Line Group 2
1003 1004
Hunt List
Hunt Pilot
1-800-555-0111
Call Hunting Components
• Hunt pilot, hunt list, and line groups
providehunting capabilities:
1st choice 2nd choice
Line Group
 Specifies the hunt option and
distribution algorithm instead
 Points to actual extensions
Hunt Pilot
 Matches dialed number for call
coverage
 Performs digit manipulation
 Points to a Hunt List for
routing
 Last-resort call forwarding
Hunt List
 Chooses path for call routing
 Points to prioritized line
groups
Endpoints
 IP phones
 Voice-mail ports

147147147
Media Resources Functions
Function
Voice termination
TDM legs must be terminated by hardware that performs
coding/decoding and packetization of the stream. This is
performed DSP resources residing in the hardware module.
Audio Conferencing
A conference bridge joins multiple participants into a single
call. It mixes the streams together and creates a unique
output stream for each connected party.
Transcoding
A transcoder converts an input stream from one codec into
an output stream that uses a different codec.
Media Termination
Point (MTP)
An MTP bridges the media streams together and allows
them to be set up and torn down independently.
Annunciator
An annunciator streams spoken messages and various call
progress tones.
Music on Hold
MOH provides music to callers when their call is placed on
hold, transferred, parked, or added to a conference.

148148148
Media Resource Matrix
Software Hardware
Voice Termination No Yes
Audio Conferencing Yes Yes
Transcoding No Yes
Media Termination Point Yes Yes
Annunciator Yes No
Music on Hold Yes No*
*SRST MOH supported

149149149
Media Resource Signaling and Audio Streams
–All media resources register with the Cisco UCM.
–Signaling between hardware media resources and Cisco UCM uses
Cisco Skinny Client Control Protocol (SCCP).
–Audio streams are always terminated by media resources.
–There are no direct IP phone-to-IP phone audio streams if a media
resources are involved.

150150150
Voice Termination Signaling and Audio Streams
–Voice termination applies to a call with a TDM and a VoIP call
leg.
–TDM leg is terminated by hardware (coding/decoding,
packetization).
–Termination is performed by DSPs installed in the gateway.
–Signaling occurs between gateway and Unified CM and between
phone and Unified CM.
PSTN
DSPs for
Voice
Termination
PSTN Call
Audio
Signaling
VoIP
TDM

151151151
Audio Conferencing Signaling and Audio
Streams
–A conference bridge joins multiple participants into a single call.
–Audio streams exist between IP phones and conference bridge and
between gateway and conference bridge.
–Signaling occurs between IP phones and Unified CM, between
conference bridge and Unified CM, and between gateway and
Unified CM.
PSTN
Conference
Call
Audio
Signaling
Integrated
Conference
Bridge

152152152
Transcoding Signaling and Audio Streams
–A transcoder converts streams from one codec into another.
–The transcoder in the example above runs in the Cisco IOS router.
–Audio streams exist between IP phones and transcoder and between
application server and transcoder.
–Signaling occurs between IP phones and Unified CM, between
transcoder and Unified CM, and between application server and
Unified CM.
PSTN
Hardware
Transcodi
ng
Applicati
on Server
Transcoded Call
Audio
Signaling
G.71
1
G.72
9
G.71
1
G.72
9

153153153
Audio Conferencing Media Resources
–Unified CM supports hardware and software conference bridges.
–The software-based conference bridge only supports single-mode
conferences, using the G.711 codec.
–Some hardware-based conference bridges support mixed-mode
conferences with participants using different codecs.
PSTN
Hardware
Conference
Bridge in
Cisco IOS
Router
Hardware Conference
Bridge in Switch
Chassis
(CMM-Module)
Software
Conference
Bridge in
Unified CM
Server

154154154
Software Audio Conferencing Bridge
–Part of Cisco IP Voice Media Streaming Application service.
–Software audio conference limitations.
•Unicast audio streams only.
•Any combination of G.711 a-law, G.711 mu-law, or wideband
audio streams may be connected.
–The maximum number of audio streams is 128* per server.
*Maximum 48 participants when Cisco UCM service is activated.
Minimum
Participants
Maximum
Participants
Default
Participants
Ad Hoc 3 64 4
Meet-Me 1 128 4

155155155
Hardware Audio Conferencing
Cisco UCM Resource Type Conferences Resource
Cisco Conference Bridge Hardware WS-X6608-T1, WS-X6608-E1
Cisco IOS Conference Bridge NM-HDV
Cisco Conference Bridge (WS-SVC-CMM) WS-SVC-CMM
Cisco IOS Enhanced Conference Bridge PVDM2, NM-HD, NM-HDV2
Cisco Video Conference Bridge (IPVC-35xx) IP/VC-35xx

156156156
Built-in Conference Resource Characteristics
–IP phones with built-in conference resources allow three-way
conferences.
–Only invoked by Barge feature.
–G.711 support only.

157157157
Meet-Me and Ad Hoc Conferencing
Characteristics
–Meet-Me
•Allocate directory numbers
•Manual distribution of Meet-Me number
•No password-like access security to enter the conference
–Basic Ad Hoc
•Conference originator controls the conference
•Originator can add and remove participants
–Advanced Ad Hoc
•Any participant can add and remove other participants
•Link multiple ad hoc conferences together

158158158
Music on Hold Media Resources
–Unified CM uses an integrated software Music on Hold server.
–For special cases, external media streaming servers can be used.
–The Unified CM integrated Music on Hold server supports
multicast and unicast for MOH streaming.
PSTN
MOH as Multicast Stream
from External Media
Streaming Server
Integrated Software MOH
Server in Unified CM
Server

159159159
Music on Hold Sources
–MOH sources
•One fixed source using a Cisco MOH USB audio sound card
•50 audio file sources
•MOH Audio File Management converts the audio file
–Codecs used for MOH are G.711, G.729, and wideband
•G.729 is developed and optimized for speech compression and
reduces the music quality
–Consider the legalities and the ramifications of rebroadcasting
copyrighted audio materials
MOH
server
Audio 1 (G.711a-
law)
Audio 1 (G.711mu-
law)
Audio 1 (G.729)
Audio 1 (Wideband)Audio 2 (G.711a-
law)
Audio 2 (G.711mu-
law)
Audio 2 (G.729)

160160160
Unicast Music on Hold
• Music on Hold unicast characteristics:
–Stream sent directly from MOH server to requesting endpoint
–Point-to-point, one-way audio stream
–Separate audio stream for each connection
–Negative effect on network throughput and bandwidth
–Unicast is useful in networks where multicast is not enabled and
devices are not capable of multicast
CM service
MOH server
IP Address
Unicast MOH
Unicast MOH

161161161
Multicast Music on Hold
• Music on Hold multicast characteristics:
–Streams sent from MOH server to a multicast group IP address
–Endpoints request an MOH audio stream and join as needed
–Point-to-multipoint, one-way audio stream
–Conserves system resources and bandwidth
–Multiple users share the same audio stream
–Networks and devices have to support multicast
–Use the multicast group IP address 239.1.1.1 to 239.255.255.255
–Increment multicast on IP address for different audio sourcesCM
service
MOH
server Multicas
t MOH
Join Multicast
Group
Multicast
Group

162162162
MOH Audio Source Selection
• The MOH stream that an endpoint receives is determined by:
–User Hold Audio Source of the device placing the endpoint on hold.
–The prioritized list of MOH resources of endpoint (holdee) placed on
hold.
–Audio sources can be configured in service parameters, device pools,
devices and the lines.
–Make sure that configured audio files are available on all TFTP servers.
Server
MOH B
Server
MOH A
Audio
1
Audio
2
Audio
3
Audio
4
Audio
1
Audio
2
Audio
3
Audio
4
Phone B
User Hold Audio 2
1. Priority MOH
Server B
Phone A
User Hold Audio 4
1. Priority MOH
Server A
Phone
B puts
Phone
A on
hold
Use MRGL A
Listen to
Audio 2

163163163
Step 1: Capacity Planning
Cisco Platform Codecs MOH Session
MCS 7815
MCS 7825
G.711a, G711u
G.729
Wideband
Co-resident or Standalone
250 MOH Streams
MCS 7835
MCS 7845
G.711a, G711u
G.729
Wideband
Co-resident or Standalone
500 MOH Streams
 The maximum of 51 unique audio sources counts for the
cluster.
 250 is the default value for unicast MOH sessions per
server.
 Each multicast MOH audio source must be counted as two
MOH streams.
 Maximum of 204 multicast streams (51 sources x 4 codec

164164164
Annunciator Overview
–The annunciator is part of the Cisco IP Voice Media Streaming
Application service.
–Annunciator streams spoken messages and various call progress
tones.
–Receiving devices such as IP phones or gateways must be capable
of SCCP to utilize this feature.
PSTN
Integrated
Annunciator
in Unified CM
server

165165165
Annunciator Features and Capacities
–Tones and announcements are predefined.
–The announcements support localization and may be customized
by replacing the appropriate .wav file.
–The annunciator is capable of supporting G.711, G.729, and
wideband codecs without any transcoding resources.
–The following features require an annunciator:
•Cisco Multilevel Precedence Preemption (call failure)
•Integration via SIP trunk (call progress and DTMF tones)
•Cisco IOS gateways and intercluster trunks (ringback)
•System messages (call failure)
•Conferencing (Barge tone)

166166166
Annunciator Performance
–A standalone server without the Cisco CallManager service can
support up to 255 simultaneous announcement streams.
–High-performance server with dual CPUs can support up to 400
announcement streams.
–Default is 48 announcement streams and recommended when co-
resident.
–Multiple standalone servers can be integrated to support the
required number of announcement streams.

167167167
The Need for Media Resource Access Control
–By default, all existing media resources usage is load-balanced.
–Usage of the hardware conference resources is preferred.
Unified
CM
Cluster
Software
Conference Bridge
SW_CFB_2
Software
Conference Bridge
SW_CFB_1
Hardware
Conference Bridge
SW_CFB_2
Hardware
Conference Bridge
SW_CFB_1
Which one
should be used
to establish a
conference?

168168168
Media Resource Design
Media
Resource
Group
List
Media
Resource
Group
Media Resource
1
Media Resource
2
Media Resource
3
Media Resource
1
first
choice
second
choice
User Needs
Media Resource
Media
Resource
Manager
Media
Resource
Group
Assigned to Device
or Device Pool
Similar to Route Lists
and Route Groups
load sharing load sharing

169169169
Common Cisco UCM User Features
–Call Park and Directed Call
Park
–Call Pickup
–Hold Reversion
–DND (Do Not Disturb)
–Intercom
–Cisco Call Back
–Barge and Privacy
–User Web Pages
–IP Phone Services
PSTN
Cisco
Unified
CM
Cluster

170170170
Call Park
–Allows you to put a call on hold so that it can be retrieved from
another telephone in the cluster.
–Can park the call to a Call Park extension by pressing the Park
softkey or the Call Park button.
–Define either a single directory number or a range of unique
directory numbers for use as call park extension numbers.
Cisco
Unified
CM
Dial
―1234‖ to
pick up
call
Call
Park
Sends Call
Park code to
display on
phone
―123
4‖
A B
C
3
2
1
5
4
Initial
stream
Call park
code
Final
stream

171171171
Directed Call Park
–Allows you to transfer a call to an available user-selected Directed
Call Park number
–Retrieve a parked call by dialing a retrieval prefix followed by the
directed call park number
–Users can also use the BLF to speed dial a Directed Call Park
number
Cisco
Unified
CM
Dial ―2180‖ or
use BLF Button
to
pick up parked
call
Transfer to
Directed Call
Park number (80)
Transfer
to 80
A B
C
3
2
1
Initial stream
Transfer to
Call Park
Final stream
4

172172172
Call Pickup and Group Call Pickup
–Call Pickup—Allows users to pick up incoming calls within their own
group.
•Cisco Unified CM automatically dials the configured call pickup
group number when the user presses Pickup.
–Group Call Pickup—Allows users to pick up incoming calls from another
group.
•After pressing Gpickup button, user must enter the appropriate
pickup group number.
Group A Group B Group C
Call Pickup Group Call
Pickup
GPickup,
dials
call
pickup
group
number
Pickup

173173173
Other Group Call Pickup
–Allows users to pick up incoming calls in a group that is associated
with their own group.
–Cisco Unified CM automatically searches for incoming calls in
associated groups when the user activates this feature.
–Use the softkey OPickup.
Group C is associated with
Group A and B
Group A Group B Group C
OPicku
p

174174174
Hold Reversion
–The Hold Reversion feature alerts a phone user when a held call
exceeds a configured time limit.
–Alerts are generated, such as a ring or beep, at the phone to
remind the user to handle the call.
Cisco
Unified
CM
A calls C
Call
Hold B
Sends Hold
Reversion
message to A
after
Timeout A B
C
3
2
1
4
Initial call
Hold Reversion
Second call
A calls
B

175175175
Do Not Disturb (DND)
–Do Not Disturb (DND) feature allows you to turn off the ringer for
an incoming call by pressing a feature button, softkey, or using the
User Options web page.
–Users can choose to have the IP phone beep or flash to indicate an
incoming call.
Cisco
Unified
CM
A
B
DND

176176176
Intercom
–With an intercom line, a user can call the intercom line of another
user, which auto-answers to one-way audio whisper.
–The recipient can then accept the whispered call and initiate a
two-way intercom call.
A B One-way audio
whisper Two-way intercom
call
 User at Phone B
receives short spoken
message of User A by
one-way audio
whisper. User B
accepts Intercom call
by pressing key. Two-
way Intercom call is
established.
 User
presses the
Intercom
button to
dial the
Intercom
line of
phone B

177177177
Barge and Privacy Overview
–Barge: Users can add themselves to remotely active calls on shared line.
•Barge uses built-in conference bridge; cBarge uses shared conference
bridge.
–Privacy: Users can allow or disallow other users on shared line to view call
information or to use Barge or cBarge.
1. Original two-party call
2. Initiator barges into the call three-way call:
– If initiator hangs up, original call remains active.
– If target hangs up, initiator and other party connect
point-to-point.
– If other party hangs up, original call and barged call
Initiator Target Other Party
Media
Barge Process
2 1
Media
Shared
line

178178178
User Options Web Page
–Controllable features vary by phone model
–Some user-definable settings are:
•User locale
•User password
•Do Not Disturb (On/Off)
•Call Forward (All, On Busy, On No Answer, On No Coverage)
•Message Waiting Indicator and Ring settings
•Line text label
•Speed dials
•IP phone services and service buttons
•Personal address book

179179179
IP Phone Services
–Cisco Unified IP Phone Services are applications that utilize the
web client or server and XML capabilities of the Cisco Unified IP
phone
–Phone service applications provide value-added services by
running directly on the user desktop phone
–Functions of a service application using IP Phone Services are
•display of data (text and graphics)
•user input
•authentication
•a mix of those functions
–Common examples for IP Phone Services are stock tickers, meal of
the day, Cisco Extension Mobility, internet news readers

180180180
Cisco Unified Presence Solutions
• Multiple options to integrate presence:
–Cisco UCM Presence
•Speed-dial presence
•Call history presence
•Presence policy
– Cisco Unified Presence Server
•User status information
•Cisco IP Phone Messenger application
•Cisco Unified Personal Communicator
•Third-Party Presence Server Integration

181181181
Cisco UCM
Presence Characteristics
–Natively supported by Cisco UCM
–Allows an interested party (a watcher) to monitor the real-time
status of a directory number (a presence entity)
–Watcher subscribes to status information of the presence entity
–Watcher can show the status of a presence entity using:
•Presence-enabled speed dials
•Presence-enabled lists (call and directory lists)
–Three possible states of watched directory number:
•Entity is unregistered
•Entity is registered—on-hook
•Entity is registered—off-hook

182182182
Cisco UCM
Presence Operation
2. Bryan’s
phone goes
off-hook
Off-hook
1. John has subscribed
for status of Bryan’s
phone
3. Information about
Bryan’s phone is sent
to John’s phone
4. John’s phone shows
Bryan’s phone in off-
hook state

183183183
Cisco UCM
Support for Presence
–Directory numbers (lines) of Cisco IP phones can be watched
•By Cisco IP phones
•By SIP devices through a SIP trunk
–Directory numbers (lines) of Cisco IP phones, and endpoints that
are reached via SIP trunks, can be watched by the following:
•Cisco IP phones
•SIP devices through a SIP trunk

184184184
Presence status can be seen on speed-dial
buttons, call lists and directories.
Watching Presence Status on Cisco IP Phones

185
Cisco IP Telephony Party 2 &
Unified Communication Troubleshooting
CIPT 2 & TUC

186186186
Course Agenda
• Multisite Deployment
• Centralized Call Processing
• Bandwidth management and Call Admission Control
• Features and Application for Multisite Deployment
• IP Telephony Security

187
Multisite Deployments
Identifying Multisite Deployment Solutions

188188188
Outline
–Multisite Deployment Solution Overview
–QoS
–Solutions to Bandwidth Limitations
–Availability
–Dial Plan Solutions
–NAT and Security Solutions

189189189
Multisite Deployment Solutions
Cisco
Unified
Communicatio
ns Manager
PSTN
Main
Site
Remote
Site
WAN
ITSP3001–
3099
3001–3099
Private
Internal
IP
Addresse
514-665-
2323
Public
IP
Network
QoS, CAC,
RTP-header
compression,
local media
resources
SRST,
PSTN
backup,
MGCP
fallback
Cisco
Unified
Border
Element
416-444-
2222
Access and
site codes,
digit
trans-
formation

190190190
Availability Options
–PSTN backup
–MGCP fallback
–Fallback for IP phones:
•SRST
•Cisco Unified Communications Manager Express
in SRST mode
–CFUR
–AAR and CFNB
–Mobility solutions:
•Extension mobility
•Device mobility
•Mobility

191191191
PSTN Backup
• Intersite calls are rerouted over the PSTN in case of an IP WAN
failure.
Cisco
Unified
Communicatio
ns Manager
PSTN
Main
Site
Remote
Site
3001–3099 3001–3099
416-555-
1234
514-555-
2222
WAN

192192192
MGCP Fallback: Normal Operation
–MGCP gateway is registered with Cisco Unified Communications
Manager over IP WAN.
–Cisco Unified Communications Manager is the MGCP Call Agent
controlling the MGCP gateway.
Cisco
Unified
Communicatio
ns Manager
Gateway
PSTN
Main
Site
Remote
Site
WAN
MGCP
control
Default
Application
(H.323 or SIP)
Gateway Fallback
MGCP
Application

193193193
MGCP Fallback: Fallback Mode
–Communication between Cisco Unified Communications Manager and
MGCP gateway is broken.
–MGCP gateway falls back to its default call-control application
(H.323 or SIP)
Cisco
Unified
Communicatio
ns Manager
Gatewa
y
PSTN
Main
Site
Remote
Site
MGCP
Application
Default
Application
(H.323 or SIP)
Gateway Fallback
WAN

194194194
Fallback for IP Phones: Normal Operation
–Remote IP phones are registered with Cisco Unified Communications
Manager over IP WAN.
–Cisco Unified Communications Manager controls IP phones.
Cisco
Unified
Communicatio
ns Manager
Gatewa
y
Remote
Gatewa
y
Main
Site
Remote
Site
Register
PSTN
WAN

195195195
Fallback for IP Phones: Fallback Mode
–Communication between Cisco Unified Communications Manager and
IP phones is broken.
–IP phones register with local gateway (either SRST or Cisco Unified
Communications Manager Express in SRST mode).
Cisco
Unified
Communicatio
ns Manager
Gatewa
y
Remote
Gatewa
y
Main
Site
Remote
Site
Register
PSTN
WAN

196196196
Using CFUR to Reach Remote-Site IP Phones Over
the PSTN During WAN Failure
• The remote site lost connectivity to main site. Phones are registered
to remote gateway:
–Main site’s Cisco Unified Communications Manager does not route calls
to the affected IP phones’ directory numbers.
–CFUR allows routing to alternate numbers for affected (unregistered) IP
phones.
Cisco
Unified
Communicatio
ns Manager
Gatewa
y
Remote
Gatewa
y
Main
Site
Remote
Site
Register
PSTN
3001
3001
unregistered
CFUR:
9-1-416-555-
3001
Direct Inward
Dialing: 416-
555-3001 to
WAN

197197197
Using CFUR to Reach Users of Unregistered
Software IP Phones on Their Cell Phones
• When a user at the main site shuts down his or her laptop with Cisco
IP Communicator:
–Main site’s Cisco Unified Communications Manager does not route calls
to the affected IP phone’s directory number.
–CFUR allows routing to alternate numbers of user (e.g., cell phone).
Cisco
Unified
Communicatio
ns Manager
Gatewa
y
Main
Site
PSTN
PC shutdown
1007
unregistered
CFUR:
9-1512-555-
1999
IP
Communicator
Home
Phone
512-555-1999
1007

198198198
AAR and CFNB
• AAR allows rerouting of calls over PSTN if not enough bandwidth for VoIP
calls:
–Alternate destination is derived from the external phone number mask and a prefix
configured per AAR group.
–Individual destinations can be configured per phone (CFNB).
Cell
Phone
512-555
-1999
Cisco
Unified
Communicatio
ns Manager
PSTN
Main
Site
Remote
Site
3001–
1099
3001–
1099
CAC Failure to IP
Phone of User X
(1009)
1009
User X
1009
configured
with CFNB:
9-1512-555-
1999
WAN

199199199
Mobility Solutions
• When users or devices roam, the resulting limitations in features can
be solved by mobility solutions:
–Device mobility
•Solves issues that result from roaming devices (region, location, SRST
reference, AAR group, CSS, etc.)
•Makes Cisco Unified Communications Manager aware of physical location of
IP phone (usually software phone such as Cisco IP Communicator)
–Extension mobility
•Solves issue of missing personal IP phone setting that results from using a
different IP phone in another office (directory number, CSS, etc.)
•Allows users to log in to IP phone and get personal configuration applied to
currently used IP phone
–Cisco Unified Mobility
•Solves issues of having different phones (office IP phone, cell phone, home
office phone, etc.)
•Allows users to be reached by a single number, independent of the phone that
is actually used

200200200
Dial Plan Solutions for Multisite Deployments
–Overlapping and nonconsecutive numbers:
•Solved by access code and (unique) site code
•Allows routing independent of directory numbers
•Appropriate digit manipulation required
–Variable-length numbering
•Dial string length determined by timeout
•Overlap sending and receiving
–DID ranges, E.164 addressing
•Use of IVR applications (AA, B-ACD, etc.) or attendant required if no
DID numbers
•Directory numbers appended to PSTN number (with variable-length
dial plans—if supported by PSTN)
–Number presentation (ISDN TON)
•Digit manipulation of incoming ISDN numbers depending on TON
–Toll bypass, TEHO, PSTN backup
•Call routing and path selection based on prioritized paths

201201201
Dial Plan Components in Multisite Deployments
Dial Plan Component Cisco IOS Gateway
Cisco Unified
Communications Manager
End point addressing
ephone-dn, dynamic
POTS, dial peers
Directory number
Call routing and path
selection
Dial peers
Route patterns, route groups,
route lists, translation patterns,
partitions, CSSs
Digit manipulation
Voice translation profiles
prefix, digit-strip, forward-
digits, num-exp
Translation patterns, route
patterns, route lists, significant
digits
Calling privileges COR and COR lists
Partitions, CSSs, time
schedules, time periods, FACs
Call coverage
Dial peers, call
applications, ephone hunt
groups
Line groups, hunt lists, hunt
pilots

202202202
Cisco Unified Border Element in
Flow-Through Mode
Cisco
Unified
Communicatio
ns Manager
Company A
Internet
Private IP Network:
10.0.0.0/8
Cisco Unified
Border
Element
ITSP
Public IP
Address A
SIP
RTP
RTP
SIPSCCP
Signaling and
media packets
repackaged
Signaling: 10.1.1.1 to 10.3.1.1
10.1.1.1
Public IP
Address B
Signaling: A (public IP) to B
(public IP)
RTP:10.2.1.5 to 10.3.1.1
10.2.1.5
RTP: A (public IP) to B (public
IP)
Private IP
Address:
10.3.1.1

203
Implementing Multisite Connections

204204204
Outline
–Examining Multisite Connection Options
–Trunk Implementation Overview
–Implementing SIP Trunks
–Implementing Intercluster and H.225 Trunks

205205205
Connection Options for
PSTN
Main
Site
IP
Remote
Site
Remote
Cluster
ITSP
Interclus
ter Trunk
MGCP
Gateway
Cisco
Unified
Border
Element

206206206
SIP Trunk Characteristics
–Distributed dial plan
–Can be conected to any device supporting SIP, including Cisco IOS
gateways, Cisco Unified Border Element, remote Cisco Unified
Communications Manager clusters, SIP network servers (proxy), etc.
–Simple, customizable protocol; rapidly evolving feature set
PSTN
Main
Site
IP
Remote
Cluster
ITSP
SIP Trunk SIP Trunk
SIP

207207207
H.323 Trunk Overview
Nongatekeeper-controlled
ICT
IP
Cisco Unified
Communications
Manager
Cluster A
Cisco Unified
Communications
Manager
Cluster C
Cisco Unified
Communications
Manager
Cluster B
Cisco Unified
Communications
Manager
Cluster D

208208208
H.323 Trunk Comparison
Nongatekeeper-
Controlled ICT
Gatekeeper-
Controlled ICT
H.225 Trunk
IP address
resolution
IP address specified
in trunk configuration
IP address resolved by H.323 RAS
(gatekeeper)
Gatekeeper call
admission
No Yes, by H.323 RAS (gatekeeper)
Scalability Limited Scalable
Peer
Cisco Unified
Communications
Manager
Prior to Cisco
CallManager 3.2
Cisco CallManager
3.2 or higher and all
other H.323 devices

209209209
Trunk Implementation Overview

210210210
Nongatekeeper Controlled ICT and SIP Trunk
Configuration Overview
• Nongatekeeper controlled ICT
and SIP trunk configuration:
–Trunk with IP address of peer
–Route pattern, route list, route group
Cisco Unified
Communications
Manager ClusterNongatekeeper-controlled ICT
IP
Cisco Unified
Communications
Manager
Cluster
SIP trunk
10.1.1.1
10.2.1.1
10.3.1.1
Access and Site
Code: 9.222
4-digit
Directory
NumbersAccess
And Site Code:
9.333
4-digit Directory
Numbers
Cisco Unified
Communications
Manager
Cluster

211211211
Gatekeeper-Controlled ICT and H.225 Trunk
Configuration Overview
• Gatekeeper-controlled ICT and
H.225 trunk configuration:
–Gatekeeper
–Trunk pointing to gatekeeper
–Route pattern, route list, route
group
IP
10.1.1
.1
10.3.1
.1
10.2.1
.1
10.9.1
.1
GK prefix:
416 GK
prefix:
409
GK prefix:
410
Cisco Unified
Communications
Manager
Cluster
Cisco Unified
Communications
Manager
Cluster
Cisco Unified
Communications
Manager
Cluster

212212212
Cisco Unified Communications Manager
SIP Trunk Configuration
Cisco Unified Communications Manager Administration: Device >
Trunk > Add New
First
choose
trunk type
and click
Next.
Enter trunk
name and
description
and choose
device
pool.

213213213
Cisco Unified Communications Manager SIP
Trunk Configuration (Cont.)
Enter IP
address of
other device
at end of SIP
trunk.
 SIP Trunk Security Profiles are used to enable and disable
security features on SIP trunks; they are configured by
navigating to System > Security Profile > SIP Trunk Security
Profile; a default profile (with security disabled) exists.
 SIP profiles are used to set timers and some feature
settings; they are configured by navigating to Device >
Device Settings > SIP Profile; a default profile exists.
SIP Trunk,
Security
Profile,
and SIP
Profile
have to be
chosen. These
are mandatory
parameters;
no default
values exist.

214214214
Implementing Intercluster and H.225 Trunks

215215215
Cisco Unified Communications Manager
Nongatekeeper-Controlled ICT Configuration
Enter trunk
name,
description,
and device
pool.
Trunk > Add New
First choose
trunk type
and click
Next.

216216216
Cisco Unified Communications Manager Nongatekeeper-
Controlled ICT Configuration (Cont.)
Enter IP address
of device on other side.

217217217
Cisco Unified Communications Manager Gatekeeper-
Controlled ICT and H.225 Trunk Configuration
Cisco Unified Communications Manager Administration: Device > Gatekeepe
Enter IP
address of
gatekeeper.
Enter
description.
Make sure
gatekeeper is
enabled.
1.Add the gatekeeper to Cisco Unified Communications Manager.
2.Add gatekeeper-controlled intercluster trunk or H.225 trunk (

218218218
Controlled ICT and H.225 Trunk Configuration (Cont.)
Enter trunk
name,
description,
and device
pool.
Trunk > Add New
Choose trunk
type and
click Next.

219219219
Controlled ICT and H.225 Trunk Configuration (Cont.)
Choose previously
configured
gatekeeper.
Trunks can register
as terminal or
gateway with the
gatekeeper. Choose
terminal type
gateway.
Enter the prefix that
should be registered
with the gatekeeper.
Enter the gatekeeper
zone in which the
trunk should be
registered.

220
A Methodology and Tools for Troubleshooting
Cisco Unified Communications Systems
Overview of Cisco
Unified Communications
Systems Troubleshooting

221221221
Publish
er
Subscrib
er
Cisco
Unity
IP
Communicator
x1001
7960 x1010
IP
Communicator
x1002
TOR
SFO
RNO
PRI
FXS
Modem/Fax
x1401
FXO
7960 x6110
IP
Communicator
x6001
IP Communicator
x1501
PC
Desktop
PC
Desktop
PC
Desktop
PSTN
FRPC
Desktop
PC
Desktop
Console
Console

222222222
Problem-Solving Model
FinishedDefine Problem
Observe Results
Utilize Process
Gather Facts
Consider Possibilities
Create Action Plan
Implement Action Plan
Problem Resolved
Document Facts
Yes
No
Start
Do
problem
symptoms
stop?

223
A Methodology and Tools for Troubleshooting
Gathering Information for Troubleshooting

224224224
–Overview
–Overview of Cisco Unified CallManager Troubleshooting
–Cisco Unified CallManager Serviceability
–Alarms
–Configuring Trace
–Dialed Number Analyzer
–Controlling Services
–Cisco Unified CallManager Real-Time Monitoring Tool
–Performance Monitor and Data Logging
–Alerts
–Trace & Log Central
–Trace Output
–Syslog Viewer
–Command-Line Interface
–Sniffer Traces
–Summary
Outline

225225225
Overview of Cisco Unified CallManager
Troubleshooting
Cisco Unified CallManager
Troubleshooting Tools:
– Cisco Unified CallManager
Serviceability
• Alarms
• Setting Trace
• CDR Analysis and Reporting
(CAR)
• Control Center
– Real-Time Monitoring Tool
• Alerts
• Viewing Trace
• Syslog Viewer
• Performance Monitoring
– CLI
Gateway Troubleshooting
Tools:
 show Commands
 debug Commands
Other Troubleshooting
Tools:
 Packet Sniffer
 www.cisco.com Tools

226226226
Cisco Unified CallManager Serviceability
http://ip_address/ccmservice

227227227
Alarms
Alarms:
 Provide run-time status of a system
 Provide notification of problem that has occurred
 Possible problem resolution may be included

228228228
Alarms—Configuration of Server and Service
• Server and Service:
Step 1. Choose Alarm > Configuration
Step 2. Choose the server
Step 3. Choose the service

229229229
Alarms—Alarm Destination and Level
• Choosing a destination:
Step 4. Check the box or boxes for your desired alarm destination.
Step 5. In the Alarm Event Level drop-down box, click the down arrow.
Step 6. Click the desired alarm event level for each of the destinations.
Step 7. To save your configuration, click the Save Button.

230230230
Alarms—Definitions
• Serviceability > Alarm > Definition

231231231
Alarms—Definitions (Cont.)
• CallManager Failure Alarm Definition

232232232
Configuring Trace—Choosing the Server and
Service
• Configuring Trace
• Selecting the server and service:
Step 1. Select the server
Step 2. Select the service

233233233
Configuring Trace—Filter Settings
• Trace filter settings:
Step 3. Choose your desired trace
fields and level.
Step 4. Choose the relevant trace fields or use
device-based tracing.

234234234
Cisco Unified CallManager Dialed Number
Analyzer
• Cisco Unified CallManager Dialed Number Analyzer

235235235
Analyzer (Cont.)
• Cisco Unified CallManager Dialed Number Analyzer—Analyzer
Screen

236236236
Analyzer (Cont.)
• Cisco Unified CallManager Dialed Number Analyzer—Analyzer
Results

237237237
Control Center—Feature Services
• Control Center—Feature Services

238238238
Cisco Unified CallManager RTMT

239239239
Cisco Unified CallManager RTMT (Cont.)
Cisco Unified CallManager
RTMT Monitoring
Categories:
 Summary
 Server
 Call Process
 Service
 Device
 CTI
 Performance

240240240
Performance Monitor
Performance Monitor
You can monitor Cisco
Unified CallManager
by choosing counters.

241241241
Perfmon Data Logging
Perfmon Data Logging
–Use as directed by TAC
–Enables collection of performance monitoring statistics
–May impact performance
System > Service Parameters > Cisco RIS Data
Collector

242242242
Alerts
Configuring Alerts:
 Preconfigured read only
 User defined

243243243
Custom Alerts on Performance Counters
Setting a custom alert on a performance counter:
Step 1. Select the counter and right-click on the selected
counter.
Step 2. Enable the alert, set the severity level, and optionally
add a custom description.

244244244
Setting a custom alert on a performance counter:
Step 3. Set the desired threshold values and when the alert
should be triggered.
Step 4. Set limits on the frequency and time that the alert can
be sent.
Custom Alerts on Performance Counters (Cont.)

245245245
RTMT - Trace and Log Central:
 Remote Browse – Browse trace and log files on a server remote from RTMT.
 Collect Files – Collect files for local viewing from remote server.
 Query Wizard – Collect and download files that match a search criteria.
 Schedule Collection – Schedule collection of trace and log files.
 Local Browse – View trace and log files that have been downloaded locally.
 Real Time Trace – View trace and log output as it is generated.
 Collect Crash Dump – Collect a core dump of trace files (TAC requested).
Trace and Log Central—Remote Browse

246246246
Trace and Log Central—Remote Browse (Cont.)
Remote Browse from RTMT:
Step 1. Click Remote Browse
Step 2. Select Trace Files

247247247
Step 3. Select the desired
Services/Applications on the
desired servers

248248248
Step 4. Select the desired
system logs on the desired
servers

249249249
Step 5 – Select the file to be
viewed.

250250250
Voice Log Translator
VLT is available as a Cisco Unified CallManager RTMT
plug-in for Cisco Unified CallManager 5.0

251251251
CLI
–Maintenance and recovery
–Locally
–Remotely using SSH
–No Linux shell access
show*
set*
delete*
unset*
file*
utils*
run*
show status
show logins
show hardware
show workingdir
show web-security
show smtp
show myself
show account
show registry
show trace
show timezone*
show cert*
show ipsec*
show version*
show packages*
show network*
show stats*
show firewall*
show process*
show tech*
show risdb*
show perf*
set timezone
set web-security
set smtp
set account
set commandcount*
set logging*
set workingdir*
set network*
set password*
set ipsec*
set cert*
set trace*
delete dns
delete smtp
delete process
delete account
delete ipsec*
unset ipsec*
unset network*
file check
file list*
file view*
file search*
file get*
file dump*
file tail*
file delete*
utils iothrottle*
utils network*
utils ntp*
utils service*
utils system*
utils remote_account*
utils disaster_recovery*
utils csa*
utils netdump*
utils dbreplication*
utils snmp*
utils sftp*
utils soap*
run sql

252252252
Sniffer Traces
–Requires third-party
software
–Capture packets
•H.323
•SIP
•SCCP
•MGCP
•RTP, SRTP
–Requires the use of
SPAN port

253
Troubleshoot Cisco Unified CallManager
Related Issues
Troubleshooting Common Endpoint Registration Issues

254254254254254254
Unified Mobility
Single Number
Reach

255255255
Cisco Unified Mobility
–Cisco Unified Mobility has two components: Mobile Connect and Mobile Voice
Access (MVA).
–With Mobile Connect, calls placed to office phones ring the office phones and
associated remote phone.
–MVA allows users to call into the enterprise from any phone and place outgoing
calls that appear to come from their office phone.
Cisco Unified
Communication
s Manager
Gateway
PSTN
Mobile Connect
lets remote and
office phones
ring
simultaneously.
Call to
office
number.
Call to
Mobile
Voice
Access
directory
number.
Mobile Voice
Access
establishes a
system to create
enterprise calls
from any
location.
Remote
Phone
Office
Phone
Customer

256256256
Cisco Unified Mobility Features
–Single (office) number for multiple devices:
•Enterprise caller ID preservation
•Single enterprise voice mailbox
–User-configurable access lists to permit or deny calling numbers
that can ring a specific remote phone
–User interface to enable or disable Cisco Unified Mobility:
•Mobile Voice Access TUI
•Cisco Unified Communications Manager user webpages
–Access to enterprise features from remote phones using DTMF:
•Softkeys can be used on phones with smart client installed.
–Call logging (CDR)

257257257
PSTN
Outside
Caller
Remote
Phone of
2001
Cisco Unified
Communication
s Manager
Mobile
Connect
Office
Phone
2001
Call to
1-511-555-
2001
Gateway
479-
555-
1555
Caller ID:
479-555-1555
408-555-
1001
511-
555-
2XXX
Outside caller calls office phone 2001 (dials 1-511-555-2001).
Mobile Connect rings office phone and remote phone.
Call is picked up at remote phone; caller ID of outside caller is
preserved at remote phone.
Mobile Connect Call Flow—Incoming Calls to
Office Phone

258258258
Cisco Unified Mobility Configuration Elements
Configuration
Element
Name
Configuration Element
Function
End User
The end user is referenced by the office phone and remote destination
profile. Mobile Connect and/or MVA must be enabled.
A maximum number of remote destinations can be configured.
Phone The office phone needs to be configured with an owner (i.e., the end user).
Remote
Destination
Profile
A virtual phone device. Per office phone number, a shared line is
configured. End user, (device) CSSs, and MOH audio sources are
specified. One or more remote destinations are added.
Remote
Destination
Associated with shared line(s) of remote destination profile. Configured
with destination number. Optionally, access lists can be applied. Mobile
Phone and Mobile Connect functions are selectively enabled.
Access List
Filters used to permit or deny incoming calls placed to the office phone
to ring a remote destination. Permitted or denied caller IDs are specified.
MVA Media
Resource
Media resource used to interact with the VoiceXML call application running
on a Cisco IOS router. Only required for MVA.

259259259
Remote
Destination
Profile
User ID
CSS
Rerouting CSS
etc.
Shared Line Between Phone and
Remote Destination Profile
Office Phone 1
MAC Address
Owner
CSS
etc.
Line1: 3001
Partition
CSS
etc.
Office Phone 2
MAC Address
Owner
CSS
etc.
Line1: 3002
Partition
CSS
etc.
Line1: 3001
Partition
CSS
etc.
Line2: 3002
Partition
CSS
etc.
Remote
Destination1
:
914168391717
Remote
Destination2:
9011971380523
0
shared line shared line
Call to
shared line
rings office
phone line
and remote
destination(s
) associated
with
corresponding
line of
remote
destination
profile.

260260260
Associate user account to a phone for
mobility

261261261
Remote Destination Profile
Extension of
The user whose
Desk phone
Will be monitored
One of the destination where phone will ring when
Some one is calling the user at their desk phone

262262262
Remote Destination Info

264264264
QoS
•L2/L3 classifications and policing
•Queuing mechanisms
•LFI
•Catalyst switch QoS

265265265
L2/L3 classifications and policing
SiSi
SiSi
SiSi
SiSi
Endpoints Access Distribution Core WAN Aggregators
TRUST BOUNDARY
1
2
3
1
2
3
Optimal Trust Boundary: Trusted Endpoint
Sub-Optimal Trust Boundary
Optimal Trust Boundary: Untrusted Endpoint

266266266
Type Data FCSPTTAG
4 Bytes
802.1Q/p
Header
PRI VLAN IDCFI
Enabling QoS in the Campus
Layer 2 Classification: 802.1p, CoS
• 802.1p user priority field also
called Class of Service (CoS)
• Different types of traffic are
assigned different CoS values
• CoS six and seven are reserved
for network use
SADASFDPream.
Ethernet Frame
1
2
3
4
5
6
7
0 Best Effort Data
Medium Priority
Data
High Priority Data
Call Signaling
Video Conferencing*
Voice Bearer
Reserved
Reserved
CoS Application
* Including Audio and Video
hree Bits Used for CoS
(802.1p User Priority)

267267267
Phone VLAN = 110
Campus QoS Considerations
Trust Boundary Extension and Operation
1 Switch and Phone Exchange CDP; Trust Boundary Is Extended to IP
Phone
2 Phone Sets CoS to 5 for VoIP and to 3 for Call-Signaling
Traffic
3 Phone Rewrites CoS from PC Port
to 0
All PC Traffic Is Reset to CoS 0
4 Switch Trusts CoS from Phone and Maps CoS  DSCP for Output
Queuing
“CoS 5 = DSCP 46”
4
1 So I Will Trust Your CoS”
“I See You’re an IP Phone,”
TRUST BOUNDARY
“Voice = 5, Signaling = 3”
2
PC Sets CoS to 5 for All Traffic3
PC VLAN =
10

268268268
Classification and Marking Design
QoS Baseline Marking Recommendations
Application
L3 Classification
DSCPPHBIPP CoS
Transactional Data 18AF212 2
Call Signaling 24CS3*3 3
Streaming Video 32CS44 4
Video Conferencing 34AF414 4
Voice 46EF5 5
Network Management 16CS22 2
L2
Bulk Data 10AF111 1
Scavenger 8CS11 1
Best Effort 000 0
Routing 48CS66 6
Mission-Critical Data 26AF31*3 3

269269269
Cisco Catalyst 3550 QoS Design
Enable switch-wide QoS.
Cat2(config)#mls qos
Modify the default CoS-to-ToS mapping table. You must setup a translation between CoS and DSCP because there
at
only 8 CoS labels and 64 possible DSCP labels. The default mapping table looks like
Cat2#show mls qos maps
Cos-dscp map:
cos: 0 1 2 3 4 5 6 7
--------------------------------
dscp: 0 8 16 24 32 40 48 56
Change the defaults so that:
• CoS 3 maps to CS3 (24)
• CoS 4 maps to AF41 (34)
• CoS 5 to EF (46)
Cat2(config)#mls qos map cos-dscp 0 8 16 24 34 46 48 56
Cat2#show mls qos maps
Cos-dscp map:
cos: 0 1 2 3 4 5 6 7
--------------------------------
dscp: 0 8 16 24 34 46 48 56

270270270
For all Signanling traffic that exceed 39K
mark down the DSCP to 8 – Cat3550
CAT2(config)#mls qos map policed-dscp 0 24 46 to 8  ! Excess traffic marked 0 or CS3 or EF will be remarked to CS1
CAT2(config)#
CAT2(config-cmap)#class-map match-all SIGNALING
CAT2(config-cmap)# match access-group name ACL_SIGNALING
CAT2(config-cmap)#exit
CAT2(config)#
CAT2(config)#policy-map VOICE-CONTROL
CAT2(config-pmap-c)#class SIGNALING
CAT2(config-pmap-c)# set ip dscp 24  ! Signaling is marked to DSCP CS3
CAT2(config-pmap-c)# police 39000 8000 exceed-action policed-dscp-transmit
CAT2(config-pmap-c)#class class-default
CAT2(config-pmap-c)# set ip dscp 0
CAT2(config)#
CAT2(config)#interface FastEthernet0/1
CAT2(config-if)# service-policy input VOICE-CONTROL
CAT2(config-if)#exit
CAT2(config)#
CAT2(config-ext-nacl)#ip access list extended ACL_SIGNALING
CAT2(config-ext-nacl)# permit tcp any any range 5000 5002
CAT2(config-ext-nacl)#end
CAT2#

271271271
Catalyst 6509 QoS
Cat6> (enable) set qos enable
QoS is enabled.
Cat6> (enable)
CaT6S> (enable) show qos status
QoS is enabled on this switch.
Trust-DSCP Workaround ACL for Catalyst 6500 2Q2T-TX/1Q4T-Rx Non-Gigabit Linecards
Cat6> (enable) set qos acl ip TRUST-DSCP trust-dscp any
Cat6> (enable) commit qos acl TRUST-DSCP
Cat6> (enable) set qos acl map TRUST-DSCP 5/2
Following command trust DSCP from end devices
Cat6(enable) set port qos 5/1 trust trust-dscp

272272272
Cisco 6509 QoS
Cat6> (enable) set qos cos-dscp-map 0 8 16 24 32 46 48 56
Cat6> (enable) set qos policed-dscp-map 0,24:8  Policed excessive traffic
Cat6> (enable)
# Create a policer that control all excessive traffic. If signalling traffic exceed 39K then drop the DSCP value from
24 to 8
Cat6> (enable) set qos policer aggregate VOIP_SIGNAL rate 39 burst 8000 policed-dscp
Cat6>(enable) set qos acl ip IPACL_NAME dscp 24 aggregate VOCE_SIGNAL tcp any any range 5000 5002
Cat6>(enable) commit qos acl IPACL_NAME
Cat6> (enable) set qos acl map IPACL_NAME 3/1

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Classify Traffic using Access-List
Cat2(config)#ip access-list extended VOICE
Cat2(config-ext-nacl)#remark Match the UDP ports that VoIP Uses for Bearer Traffic
Cat2(config-ext-nacl)#permit udp any any range 16384 32767
Cat2(config)#ip access-list extended VOICE-CONTROL
Cat2(config-ext-nacl)#remark Match VoIP Control Traffic
Cat2(config-ext-nacl)#remark SCCP
Cat2(config-ext-nacl)#permit tcp any any range 5000 5002
Cat2(config-ext-nacl)#remark H323 Fast Start
Cat2(config-ext-nacl)#permit tcp any any eq 1720
Cat2(config-ext-nacl)#remark H323 Slow Start - Verify could be in 3000 range for CM or 11000 to 65535 with newer IOS's
Cat2(config-ext-nacl)#permit tcp any any range 11000 11999
Cat2(config-ext-nacl)#remark H323 MGCP
Cat2(config-ext-nacl)#permit udp any any eq 2427
Cat2(config-ext-nacl)#permit tcp any any eq 2428

274274274
WAN Edge QoS Design Considerations
QoS Requirements of WAN Aggregators
WAN Aggregator
WAN Edges
Campus
Distribution/Core
Switches
LAN Edges
Queuing/Dropping/
Shaping/Link-Efficiency
Policies
for Campus-to-Branch Traffic
WAN

275275275
WAN Edge QoS Design Considerations
Link-Speed Considerations
• Slow-speed links (≤ 768 kbps)
–Voice or video (not both)—3 to 5 class model
–LFI mechanism required
–cRTP recommended
• Medium-speed links (≤ T1/E1)
–Voice or video (not both)—5 Class model
–cRTP optional
• High-speed links (> T1/E1)
–Voice and/or video—5 to 11 Class (QoS baseline) model
–Multiple links require bundling or load-balancing
–Very high-speed links (DS-3/OC-3) require newer CPUs

276276276
WAN Edge Bandwidth Allocation Models
Three-Class (VoIP and Data Only) WAN Edge Model
Voice
33%
Call-
Signaling
5%
Best
Effort
(62%)

277277277
WAN Edge Bandwidth Allocation Models
Three-Class WAN Edge Model Configuration Example
!
class-map match-all VOICE
match ip dscp ef ! IP Phones mark Voice to EF
class-map match-any CALL-SIGNALING
match ip dscp cs3 ! Call-Signaling marking (new)
match ip dscp af31 ! Call-Signaling marking (old)
!
!
policy-map WAN-EDGE
class VOICE
priority percent 33 ! Recommended to keep LLQ ≤ 33%
compress header ip rtp ! Optional: Enables Class-Based cRTP
class CALL-SIGNALING
bandwidth percent 5 ! Minimal BW guarantee for Call-Signaling
class class-default
fair-queue ! All other data gets fair-queuing
!

278278278
Frame Relay QoS Design
FRTS (+ FRF.12) Recommended Parameters Table
PVC
Speed
CIR Bc
Fragment
Size
56 kbps 53500 bps 532 bits per Tc 70 Bytes

279279279
Frame Relay QoS Design
Slow-Speed Frame Relay Configuration Example
<a 3 to 5 Class Model can be used>
Optional: Enabling Class-Based cRTP
!
policy-map MQC-FRTS-768
class class-default
shape average 729600 7296 0 ! CIR=95% rate, Bc=CIR/100, Be=0
service-policy WAN-EDGE ! Queues packets before shaping
!
!
interface Serial2/0
no ip address
encapsulation frame-relay
!
interface Serial2/0.12 point-to-point
frame-relay interface-dlci 102
class FR-MAP-CLASS-768 ! Binds the map-class to the FR DLCI
!
!
map-class frame-relay FR-MAP-CLASS-768
service-policy output MQC-FRTS-768 ! Attaches MQC policies to FR map-class
frame-relay fragment 960 ! Enables FRF.12
!
WAG
BR
FR Link ≤ 768 kbps
Frame
Relay
Cloud

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MLPoFR QoS Design
Slow-Speed FR SIW Configuration Example at the Branch
<a 3 to 5 Class Model can be used>
Optional: Enabling Class-Based cRTP
!
interface Serial6/0
no ip address
encapsulation frame-relay
frame-relay traffic-shaping
!
interface Serial6/0.60 point-to-point
bandwidth 256
frame-relay interface-dlci 60 ppp Virtual-Template60 ! Enables MLPoFR
class FRTS-256kbps ! Binds the map-class to the FR DLCI
!
interface Virtual-Template60
bandwidth 256
ip address 10.500.60.2 255.255.255.252
service-policy output WAN-EDGE ! Attaches MQC policy to map-class
ppp multilink
ppp multilink fragment-delay 10 ! Enables MLP fragmentation
ppp multilink interleave ! Enables MLP interleaving
!
map-class frame-relay FRTS-256kbps
frame-relay cir 243500 ! CIR is set to 95% of FR DLCI rate
frame-relay bc 2432 ! Bc is set to CIR/100
frame-relay be 0 ! Be is set to 0
frame-relay mincir 243500 ! MinCIR is set to CIR
no frame-relay adaptive-shaping ! Adaptive shaping is disabled
!
WAG
BR
Frame
Relay
Cloud
MLPoFR Link ≤ 768
kbps

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