Ccvp plus module 2
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1 CCVP Plus Bootcamp Module 2
description
Summary book of CCVP
Transcript of Ccvp plus module 2
1
CCVP Plus Bootcamp
Module 2
2
Cisco IP Telephony Part 1
Faisal H. Khan
CCIE Voice Instructor
3 3 3
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
4 4 4
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
5 5 5
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)
6 6 6
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
7 7 7
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)
8 8 8
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
9 9 9
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.
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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
11 11 11
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).
12 12 12
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
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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
14 14 14
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
15 15 15
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
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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
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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.
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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
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Installation
CIPT 1
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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.
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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.
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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.
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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.
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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.
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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) Publisher
DMA
Cisco Unified
CM Release
6.0(1)
installation
imports file.
DMA exports
TAR file or
tape.
Network Share
Server
S/FTP
Appliance
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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.
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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)
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Web interface for administration
Introducing VoIP
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Cisco UCM Administration and User Interface Options
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Cisco UCM Administration CLI Main Page
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Initial Configurations
CIPT 1
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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
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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
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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.
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Managing user account
CIPT 1
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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
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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.)
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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.
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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
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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 Privilege User 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
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LDAP Integration
CIPT 1
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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
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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.
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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
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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.
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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.
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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
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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
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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.
50 50 50
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.
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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.
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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.
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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.
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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.
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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.
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Device Settings
Device Settings contain default
settings, profiles, templates, and
common device configurations that
can be assigned to a device or device
pool.
57 57 57
Device Defaults Configuration
Use device defaults to set the default
characteristics of each type of device that
registers with a Cisco UCM.
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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
template assigned.
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Softkey Template
Softkey template configuration allows the
administrator to configure softkey layouts which are
assigned to Cisco Unified IP phones.
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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.
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Common Phone Profile
Common phone profiles include phone
configuration parameters and are assigned to
IP phones.
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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
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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
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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
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Cisco SCCP IP Phone Startup Process
Unified CM Cisco TFTP DHCP
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
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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 TFTP DHCP
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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.
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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
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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.
70 70 70
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.
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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.
72 72 72
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
73 73 73
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
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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.
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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
76 76 76
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
79 79 79
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)
80 80 80
Endpoint Identifiers
MGCP PSTN/
PBX
T1/E1 VWIC
2/1/1
FXS VWIC
2/1/1
AALN/S2/SU1/[email protected]
amp.com
S1/SU1/DS1-
AALN/S2/SU1/[email protected]
amp.com Endpoint
type
(analog
line)
Slot
2
Subunit
1
Port
1
S1/SU1/DS1-
Slot 1 Subunit
1
Port
1
Endpoint type
(T1/E1 trunk)
Hostname
Hostname
81 81 81
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
82 82 82
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
83 83 83
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.921 TCP
Q.931
Q.921
CUCM Gateway PSTN CO
84
Enabling Single-Site On-Net Calling
85 85 85
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 3005 3004
Cisco
Unified CM Cisco
Unity
86 86 86
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 2003 2002 2004
PSTN
416-555-
4001
dials 4001
87 87 87
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
88 88 88
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
89 89 89
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
90 90 90
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
91 91 91
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.
92 92 92
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
93 93 93
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
94 94 94
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
95 95 95
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
96 96 96
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
97 97 97
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
98 98 98
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
99 99 99
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
100 100 100
User Input on Type A SIP Phones – SIP Dial Rules Configured on the Phone
–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‖
Call in progress, call connected, call denied, etc.
Existing SIP
phone
such as 7940,
7960
Signaling
Dialing actions:
20 0 1 Dial
Pattern 1…
Timeout 0
101 101 101
User Input on Type B SIP Phones – No SIP Dial Rules Configured on the Phone
–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,
Call in progress, call connected, call denied, etc.
SIP enhanced
phone
such as 7971
Signaling
Dialing actions:
2 0 0 1 Dial
102 102 102
User Input on Type B SIP Phones – SIP Dial Rules Configured on the Phone
–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‖
Call in progress, call connected, call denied, etc.
Signaling
Dialing actions:
2001 Dial
Pattern 2…
Timeout 0
SIP enhanced
phone
such as 7971
103 103 103
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?
104 104 104
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
105 105 105
Route
Pattern
Route
List
Route
Group
Second
Choice
Route
Group
First Choice Second
Choice
Configuration Order 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)
PSTN IP 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
106 106 106
Route List
User dials
914165265000
PSTN Route 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
107 107 107
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
108 108 108
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
109 109 109
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
110 110 110
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 #)
111 111 111
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
112 112 112
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.
113 113 113
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
114
Digit Manipulation
115 115 115
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
116 116 116
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 Use Called Party Transformation in Translation Pattern, or use Significant Digits
Overlapping endpoint directory number
Internal to Internal
PSTN to Internal
Use Called Party Transformation in Translation Pattern
117 117 117
PSTN 1005
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
118 118 118
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
119 119 119
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
– Part of Calling/Called Transformation settings.
Transformation Masks Manipulate the dialed digits or calling party number
– Part of Calling/Called Transformation settings.
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.
120 120 120
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.
121 121 121
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 option
122 122 122
Digit Prefix
–Prepend digits to the pattern
–Valid entries include the digits 0 through 9, *, and #
–Part of Calling/Called Transformations settings
123 123 123
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
124 124 124
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)
125 125 125
Using PreDot DDIs
PBX
Unified CM Match: 9.8XXX
Discard: PreDot
Called Party: 8123
User Dials: 98123
126 126 126
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
127 127 127
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.
128 128 128
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
√
129 129 129
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
130
Class of Service
131 131 131
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
132 132 132
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
133 133 133
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
134 134 134
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.
135 135 135
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
136 136 136
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).
137 137 137
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_TP
RegEmployees_TS
CiscoAustin_PT RegEmployees_TS
Start–End Repetiti
on
Time
Periods
Time Schedule
Time Schedule
Time
Periods
138 138 138
Time-of-Day Routing Configuration Procedure
1. Create time periods.
2. Create time schedules.
3. Assign time schedules to partitions.
139 139 139
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
140 140 140
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
141 141 141
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
143 143 143
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)
144 144 144
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
145 145 145
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.
146 146 146
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
147 147 147
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.
148 148 148
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
149 149 149
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.
150 150 150
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
151 151 151
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
152 152 152
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
153 153 153
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
154 154 154
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
155 155 155
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
156 156 156
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.
157 157 157
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
158 158 158
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
159 159 159
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)
Audio 2 (Wideband)
160 160 160
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
161 161 161
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 sources CM
service
MOH
server Multicas
t MOH
Join Multicast
Group
Multicast
Group
162 162 162
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
163 163 163
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
types).
164 164 164
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
165 165 165
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)
166 166 166
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.
167 167 167
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?
168 168 168
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
169 169 169
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
170 170 170
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
171 171 171
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
172 172 172
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
173 173 173
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
174 174 174
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
175 175 175
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
176 176 176
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
177 177 177
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
are released.
Initiator Target Other Party
Media Barge Process
2 1
Media
Shared
line
178 178 178
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
179 179 179
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
180 180 180
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
181 181 181
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
182 182 182
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
183 183 183
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
184 184 184
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
186 186 186
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
188 188 188
Outline
–Multisite Deployment Solution Overview
–QoS
–Solutions to Bandwidth Limitations
–Availability
–Dial Plan Solutions
–NAT and Security Solutions
189 189 189
Multisite Deployment Solutions
Cisco
Unified
Communicatio
ns Manager
PSTN
Main
Site
Remote
Site
WAN
ITSP 3001–
3099
3001–3099
Private
Internal
IP
Addresse
s
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
190 190 190
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
191 191 191
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
192 192 192
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
193 193 193
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
194 194 194
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
195 195 195
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
196 196 196
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
1003
WAN
197 197 197
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
198 198 198
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
199 199 199
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
200 200 200
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
201 201 201
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
202 202 202
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
SIP SCCP
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
Multisite Deployments
Implementing Multisite Connections
204 204 204
Outline
–Examining Multisite Connection Options
–Trunk Implementation Overview
–Implementing SIP Trunks
–Implementing Intercluster and H.225 Trunks
205 205 205
Connection Options for Multisite Deployments
PSTN Main
Site
IP
Remote
Site
Remote
Cluster
ITSP
Interclus
ter Trunk
MGCP
Gateway
Cisco
Unified
Border
Element
206 206 206
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
207 207 207
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
208 208 208
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
209 209 209
Trunk Implementation Overview
210 210 210
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 Cluster Nongatekeeper-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
Numbers Access
And Site Code:
9.333
4-digit Directory
Numbers
Cisco Unified
Communications
Manager
Cluster
211 211 211
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
212 212 212
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.
213 213 213
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.
214 214 214
Implementing Intercluster and H.225 Trunks
215 215 215
Cisco Unified Communications Manager Nongatekeeper-Controlled ICT Configuration
Enter trunk
name,
description,
and device
pool.
Cisco Unified Communications Manager Administration: Device >
Trunk > Add New
First choose
trunk type
and click
Next.
216 216 216
Cisco Unified Communications Manager Nongatekeeper-Controlled ICT Configuration (Cont.)
Enter IP address
of device on other side.
217 217 217
Cisco Unified Communications Manager Gatekeeper-Controlled ICT and H.225 Trunk Configuration
Cisco Unified Communications Manager Administration: Device > Gatekeeper > Add New
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 (see next pages).
218 218 218
Cisco Unified Communications Manager Gatekeeper-Controlled ICT and H.225 Trunk Configuration (Cont.)
Enter trunk
name,
description,
and device
pool.
Cisco Unified Communications Manager Administration: Device >
Trunk > Add New
Choose trunk
type and
click Next.
219 219 219
Cisco Unified Communications Manager Gatekeeper-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
221 221 221
Cisco Unified Communications Systems
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
FR PC
Desktop
PC
Desktop
Console
Console
222 222 222
Problem-Solving Model
Finished Define 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 Cisco Unified Communications Systems
Gathering Information for Troubleshooting
224 224 224
–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
225 225 225
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
226 226 226
Cisco Unified CallManager Serviceability
http://ip_address/ccmservice
227 227 227
Alarms
Alarms:
Provide run-time status of a system
Provide notification of problem that has occurred
Possible problem resolution may be included
228 228 228
Alarms—Configuration of Server and Service
• Server and Service:
Step 1. Choose Alarm > Configuration
Step 2. Choose the server
Step 3. Choose the service
229 229 229
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.
230 230 230
Alarms—Definitions
• Serviceability > Alarm > Definition
231 231 231
Alarms—Definitions (Cont.)
• CallManager Failure Alarm Definition
232 232 232
Configuring Trace—Choosing the Server and Service
• Configuring Trace
• Selecting the server and service:
Step 1. Select the server
Step 2. Select the service
233 233 233
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.
234 234 234
Cisco Unified CallManager Dialed Number Analyzer
• Cisco Unified CallManager Dialed Number Analyzer
235 235 235
Cisco Unified CallManager Dialed Number Analyzer (Cont.)
• Cisco Unified CallManager Dialed Number Analyzer—Analyzer Screen
236 236 236
Cisco Unified CallManager Dialed Number Analyzer (Cont.)
• Cisco Unified CallManager Dialed Number Analyzer—Analyzer Results
237 237 237
Control Center—Feature Services
• Control Center—Feature Services
238 238 238
Cisco Unified CallManager RTMT
239 239 239
Cisco Unified CallManager RTMT (Cont.)
Cisco Unified CallManager
RTMT Monitoring
Categories:
Summary
Server
Call Process
Service
Device
CTI
Performance
240 240 240
Performance Monitor
Performance Monitor
You can monitor Cisco
Unified CallManager
by choosing counters.
241 241 241
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
242 242 242
Alerts
Configuring Alerts:
Preconfigured read only
User defined
243 243 243
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.
244 244 244
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.
Step 5. Optionally, set the alert to generate an e-mail and click
Activate.
Custom Alerts on Performance Counters (Cont.)
245 245 245
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
246 246 246
Trace and Log Central—Remote Browse (Cont.)
Remote Browse from RTMT:
Step 1. Click Remote Browse
Step 2. Select Trace Files
247 247 247
Trace and Log Central—Remote Browse (Cont.)
Remote Browse from RTMT:
Step 3. Select the desired
Services/Applications on the
desired servers
248 248 248
Trace and Log Central—Remote Browse (Cont.)
Remote Browse from RTMT:
Step 4. Select the desired
system logs on the desired
servers
249 249 249
Trace and Log Central—Remote Browse (Cont.)
Step 5 – Select the file to be
viewed.
250 250 250
Voice Log Translator
VLT is available as a Cisco Unified CallManager RTMT
plug-in for Cisco Unified CallManager 5.0
251 251 251
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
252 252 252
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
254 254 254 254 254 254
Unified Mobility Single Number Reach
255 255 255
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
256 256 256
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)
257 257 257
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
258 258 258
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.
259 259 259
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.
260 260 260
Associate user account to a phone for mobility
261 261 261
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
262 262 262
Remote Destination Info
263 263 263
QOS
264 264 264
QoS
•L2/L3 classifications and policing
•Queuing mechanisms
•LFI
•Catalyst switch QoS
265 265 265
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
266 266 266
Type Data FCS PT TAG
4 Bytes
802.1Q/p
Header PRI VLAN ID CFI
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
SA DA SFD Pream.
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
Three Bits Used for CoS
(802.1p User Priority)
267 267 267
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”
“CoS 3 = DSCP 24”
“CoS 0 = DSCP 0”
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 Traffic 3
PC VLAN =
10
268 268 268
Classification and Marking Design QoS Baseline Marking Recommendations
Application L3 Classification
DSCP PHB IPP CoS
Transactional Data 18 AF21 2 2
Call Signaling 24 CS3* 3 3
Streaming Video 32 CS4 4 4
Video Conferencing 34 AF41 4 4
Voice 46 EF 5 5
Network Management 16 CS2 2 2
L2
Bulk Data 10 AF11 1 1
Scavenger 8 CS1 1 1
Best Effort 0 0 0 0
Routing 48 CS6 6 6
Mission-Critical Data 26 AF31* 3 3
269 269 269
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
270 270 270
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#
271 271 271
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
272 272 272
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
273 273 273
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
274 274 274
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
275 275 275
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
276 276 276
WAN Edge Bandwidth Allocation Models Three-Class (VoIP and Data Only) WAN Edge Model
Voice
33%
Call-
Signaling
5%
Best
Effort
(62%)
277 277 277
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
!
278 278 278
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
64 kbps 60800 bps 608 bits per Tc 80 Bytes
128 kbps 121600 bps 1216 bits per Tc 160 Bytes
256 kbps 243500 bps 2432 bits per Tc 320 Bytes
384 kbps 364800 bps 3648 bits per Tc 480 Bytes
512 kbps 486400 bps 4864 bits per Tc 640 Bytes
768 kbps 729600 bps 7296 bits per Tc 960 Bytes
279 279 279
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
280 280 280
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
