03 Cucm

Cisco Unified
Communications Manager
Presentation_ID 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 1
Agenda
1. Clustering Cisco CallManagers
2. Deployment Models
3. CallManager Signaling Architecture
4. H.323 Gateway
5. Media Gateway Control Protocol (MGCP)
6. Type of voice interfaces
7. Route Plan Basics
8. Telephony Class of Service
9. Media Resources
10. Other CallManager Features
11. Call Admission Control (CAC) and Automated Alternate Routing (AAR)
12. Survivable Remote Site Telephony (SRST)
13. Extension Mobility
14. Cisco IP Communicator
15. Cisco VT Advantage
16. Troubleshooting Tools
1. Clustering Cisco
CallManagers
Cluster Definition
SQL Database relationship
defines the cluster
Cluster has one Publisher
Publisher
server and N number of
Subscriber servers
One database on the Publisher
replicated to Subscribers
Subscriber Subscriber
Subscriber
Intra-Cluster Communication
SQL
SQL Intra-Cluster
Intra-Cluster
Database
Database Run-Time
Run-TimeData
Datain
inSDL
SDL
Publisher
Subscriber Subscriber
Subscriber
Device
DeviceRegistration
Registration
Cluster
ClusterDetermination
Determination and
andRedundancy
Redundancy
Database Writes/Updates
1. The Publisher server always Publisher
updates its own local
database for changes.
1
Client SQL
2. The Subscriber servers only
update the Publishing
servers database for
2
changes. Subscriber Subscriber
3. If Publishers database is not
available, the updates from
subscriber will not be saved. Client SQL Client SQL
X3
Datebase Client Computer
Updates
Client
Database Replication
Publisher
Changes made to the database

are replicated out via Client SQL
transactional process from
Publisher to Subscribers.
If the link is down, SQL keeps a
transaction log and replicates Subscriber Subscriber
the data when possible.
Client SQL Client SQL
Database Client Computer

Replication
Client
Intra-cluster Run Time Data: Registration
Device with extension 1000

Skinny Connect is registered to me (CCM B).
Register (active) B
TCP KA
1000
TCP Connect D
(Standby)
Cisco
CiscoCallManager
CallManagerList
List
11BB H
22DD
33HH
Clustering Options
Distribute IP phones based
Primary on DN
CallManager redundancy
Secondary/Backup groups based on DN range
Tertiary/Last Resort
To 7,500 IP Phones To 15,000 IP Phones To 30,000 IP Phones
Publisher and Publisher and

TFTP Server TFTP Server
1 to
Backup
7500
Publisher, Backup
and TFTP Server 7501 to
Backup 1 to 7500
15000
Primary CM
1 to 7500 7501 - 15000 15001 to
Backup 22500
22501 to
30000
Presentation_ID 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 9 III
Cluster Recommendations
Up to 7,500 IP Phones
A cluster of two Cisco CallManagers
Publisher is also the Secondary.
Single Active Cisco CallManager
DFWCLSTR1
Cluster
Cluster
A Publisher
Cisco
CiscoCallManager
CallManager
Group Primary CallManager
GroupBABA B
Phone 1 - 7500
A cluster of four Cisco CallManagers

Stand-alone Publisher.
Two Primary Cisco CallManagers (B and C).
One Backup Cisco CallManager (D).
A Cluster
Cluster
Publisher
B C
Cisco
CiscoCallManager
CallManager Cisco
CiscoCallManager
CallManager
Group
GroupBDBD D Group
GroupCDCD
A cluster of eight Cisco CallManagers
Stand-alone Publisher and Cisco TFTP
Four Primary Cisco CallManagers (B-C-E-G).
Two Backup Cisco CallManager (D and H).
A F
Cluster
Cluster
Publisher Cisco TFTP
Cisco B C E G Cisco
Cisco Cisco
CallManager
CallManager CallManager
CallManager
Group
GroupBDH
BDH Group
GroupEHD
EHD
Cisco
Cisco Cisco
Cisco
D H
CallManager
CallManager
Group
GroupCDH
CDH Group
GroupGHD
GHD
2. Deployment Models
Single Site: Overview
Cisco CallManager,
Applications and
DSP Resources at same physical
location Applications
(VMail, IVR, ICD, ...)
Supports up to 30,000
IP Phones per cluster
CallManager
Multiple clusters can Cluster
be interconnected
via Inter-Cluster
trunks
PSTN used for
all external calls
PSTN
Centralized Call Processing: Overview
Applications
(VMail, IVR, ICD, ...) SRST-enabled
PSTN router
CallManager
Cluster
Branch A
IP WAN
Headquarters
CallManager at central site
Applications and DSP resources can be
centralized or distributed
Supports up to 30,000 IP phones per cluster Branch B
Call admission control (limit number of calls per site)
Survivable Remote Site Telephony for remote branches
Distributed Call Processing: Overview
CallManager
Applications Cluster
(VMail, IVR, ICD, ...)
Applications
CallManager
Cluster
PSTN
Branch A
GK
IP WAN
Headquarters CallManager
Gatekeeper
Cluster
Applications
CallManager and Applications located
at each site
Up to 30,000 IP Phones per site
100+ sites
Transparent use of PSTN if IP WAN unavailable Branch B
3. CallManager
Signaling Architecture
H.323 to Skinny Client Call Flow Diagram
CallManager
H.323 SCCP
V
RTP
H.323 Cisco
PSTN Gateway IP Phone
H.323 Client to Skinny Client Call Flow
H.323 H.323 Cisco SCCP Cisco
Gateway CallManager IP Phone
H.225 Setup
H.225 Setup Ack Station Call Info
Station Set Lamp (Blink)
Station Set Ringer (On)
H.225 Alerting
Station Off Hook
H.225 Connect
Station Set Lamp (Steady)
H.245 Master/Slave Determination Station Set Ringer (Off)
H.245 Master/Slave Determination Ack

H.245 Terminal Capabilities Set
H.245 Terminal Capabilities Set Ack
H.245 Open Logical Channel

Station Start Media Reception
H.245 Open Logical Channel Ack Station Start Media Transmission
Conversation
Station On Hook
H.245 Request Channel Close
Station Stop Media Transmission
H.245 Request Channel Close Ack
Station Stop Media Reception
H.225 Release Complete Station Set Lamp (Off)
Typical Q.931 Messages
Calling Network Called
Party Party
Setup
Setup Acknowledge
Information
Call Proceeding
Setup
Call Proceeding
Alerting
Alerting Connect
Connect
Connection Acknowledge Connection Acknowledge
Cisco IP Phone Client to Cisco IP Phone
Calling
Cisco IP Phone SCCP Cisco CallManager SCCP Cisco IP Phone
Station Off Hook
Station Play Tone (Dial Tone)
Station Set Lamp (Steady)
Station Digit Dialed
Station Stop Tone (Dial Tone)
Station Digit Dialed
Station Digit Dialed Station Call Information
Station Play Tone (Ringback) Station Set Lamp (Blink)
Station Set Ringer (On)
Station Off Hook

Station Play Tone (Off) Station Set Lamp (Steady)
Station Set Ringer (Off)
Station Start Media Transmission Station Start Media Transmission
Station Start Media Reception Station Start Media Reception
Conversation
Station On Hook
Station Stop Media Transmission Station Stop Media Transmission
Station Stop Media Reception Station Stop Media Reception
Station Set Lamp (Off) Station Set Lamp (Off)
Station On Hook
4. H.323 Gateway
IOS Configuration - Sample
Dial-peer voice 10 pots
Destination-pattern 1030
Port 3/0/0
Dial-peer voice 100 voip

Destination-pattern 1 . . .
Session target ipv4:10.0.0.2
Preference 0
Dial-peer voice 101 voip

Destination-pattern 1 . . .
Session target ipv4:10.0.0.1
Preference 1
CallManager H.323 Configuration - 1
5. Media Gateway
Control Protocol
(MGCP)
General Concepts
MGCP is, in essence, a master/slave protocol, where the

gateways are expected to execute commands sent by
the Call Agents.
The call control "intelligence" is outside the gateways
and handled by external call control elements called Call
Agents.
MGCP is a text based protocol with plain-text commands
sent over UDP port 2427 called Signaling Plane.
General Concepts Call Agents
The Call Control elements that play the Master role in MGCP
are known as Call Agents.
CA provides centralized call intelligence.
A Call Manager controlling an MGCP gateway is an example of
a Call Agent.
MGCP does not define a mechanism for synchronizing Call
Agents. They need to have their own means of synchronizing.
This function is achieved in Ciscos Call Manager Clusters by
the SDL layer.
General Concepts End Points
An end point is a network element that provides
conversion between the audio signals carried on
telephone circuits and data packets carried over packet
networks.
One telephony gateway can have multiple endpoints. e.g.
Multiple ports on a c2600.
General Concepts - Connections
A point to point connection is an association between two

endpoints with the purpose of transmitting data between
these endpoints.
For point-to-point connections, the endpoints of a
connection could be in separate gateways or in the same
gateway.
Voice traffic is carried over IP/RTP called Data Plane.
General Concepts - Endpoint Identifiers
Endpoint identifiers have two components:
Domain name of the gateway that is managing the endpoint (case
sensitive)
Local name within that gateway (case insensitive)
e.g. AALN/S1/SU0/0@AV-VG200-2.cisco.com
AALN= Analog Access Line eNdpoints
S1/SU0/0 = Slot 1/SubUnit 0/Port 0
For example, MGCP FXS Call
NTFY O: L/hd
MGCP Gateway
e.g. VG200
RQNT R: L/hu,D/[0-9*#] S:dl
NTFY O: 4
CallManager
RQNT R: L/hu, D/[0-9*#] S:
NTFY O: 5
CRCX
200 OK (RTP Port Info)
MDCX
Device Requirements for Call Preservation
Active Connection
Maintenance
RTP streams between
devices
Disconnect Supervision
End user
Timed
Media Streaming Failure
(MSF)
Switchover Algorithm
Graceful
Immediate
CallManager MGCP Configuration - 1
In the upper right, select Add a New Gateway
Select the gateway from the pull down
1. Enter MGCP Domain Name (Need to match hostname and ip domain-name)
2. Description if necessary.
3. Select Module Type and Click Insert
1. Select the VIC cards used in the network module.
2. Select Module Type and Click Update.
Select the endpoints to configure them
Select the DN to configure them
IOS MGCP Configuration
mgcp
Starts the MGCP process
mgcp call-agent <primary CM ip address>
Primary CM IP address or DNS name
mgcp dtmf-relay codec all mode out-of-band
Enable DTMF relay
ccm-manager MGCP
Initializes the MGCP process (TCP Socket & MGCP Link monitor)
ccm-manager redundant-host <1st backup> <2nd backup>
Specifies the IP address or DNS name of first and second call
manager backup systems.
IOS MGCP Configuration (Contd)
ccm-manager switchback <mode>
Controls how the gateway attempts to return management to the

next higher priority Call Manager in list. (CM Systems must have
an open TCP socket to be considered viable)
Mode Configuration Options:

Not specified (default) Means no switchback; Viola ignores
return of a higher priority CM.
graceful -- Switches back after last active call ends.
immediate -- Switches back as soon as TCP link to primary is
established.
IOS MGCP Configuration (Contd)
More Switchback Options:

ccm-manager switchback schedule-time <hh:mm>
Allows specific time-of-day switchback. (If the specified
time earlier than current time, waits until next day.)
ccm-manager switchback uptime-delay <1-1440>
Allows user to set a switchback delay timer in minutes
Switchback can be forced from enable mode:
VG200-1# ccm-manager switchover-to-backup
IOS MGCP Configuration - Sample
Required MGCP statements:

mgcp <== Starts the MGCP process
mgcp call-agent <ip address #1> <== Primary Call Manager DNS name or IP address
mgcp dtmf-relay codec all mode out-of-band <== Enable DTMF relay
Required CCM-Manager statements:

ccm-manager mgcp
For Redundant configurations:

(First backup CM) (Second backup CM)
ccm-manager redundant-host <ip address #2> <ip addr #3>
Required Dial-Peers: (ONLY the following)

dial-peer voice xxx pots
application MGCPAPP <== Required!
port x/x/x
Do not use these commands
For MGCP managed endpoints (those with

application MGCPAPP in the dial-peer statement),
all dial-plan related configuration elements are
controlled by Call Manager-
Dont configure:
destination-pattern
session target
number expansion
connection PLAR/tie-line/trunk (voice-port)
Codec
IOS MGCP Config Server
Automatically download your MGCP gateway

configuration from CallManager/TFTP server:
Router(config)# ccm-manager config server 10.0.0.1

Router(config)# ccm-manager config
* But you still need to setup MGCP in the CallManager first.
IOS MGCP Config Server Result
!
ccm-manager mgcp
ccm-manager music-on-hold
ccm-manager config server 10.0.0.1
ccm-manager config
!
mgcp
mgcp call-agent 10.0.0.2 2427 service-type mgcp version 0.1
mgcp rtp unreachable timeout 1000 action notify
mgcp package-capability rtp-package
mgcp package-capability sst-package
no mgcp package-capability fxr-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp fax t38 inhibit
mgcp rtp payload-type g726r16 static
!
mgcp profile default
!
!
dial-peer voice 999300 pots
application mgcpapp
port 3/0/0
!
6. Type of voice
interfaces
7. Route Plan Basics
Dial Plan Visual Objective
Users required to dial

San Jose Secondary Voice Path
Pre-pend 1408 and send to PSTN
7 digits for Intersite calls
526-1111
Gatekeeper(s)
A
A
V
PSTN
Philadelphia
(215) 555-XXXX
(408) 526-XXXX 5 Digit Internal Dialing
5 Digit Internal Dialing
Primary Voice Path
IP WAN Strip 52 and deliver
61111 to remote Call Manager
Route Plan Call Types
Route
Pattern 9.1408XXXXXXX Gatekeeper
CallManager Remote
GK
CallManager
1000 IP WAN
Router/GW PSTN
1001 914085264000
CallManager routes two basic call types:

On-Cluster Calls Destination Directory Number (DN)
is registered with CallManager
Off-Cluster Calls External route patterns must be
configured on CallManager
Presentation_ID 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 55 I
External Route Elements in CallManager
Route Pattern Route
Matches dialed number for external calls Pattern
Performs digit manipulation (optional)
Points to a Device / Route List for routing
Configuration Order
Route
Route List List
1st 2nd
Chooses path for call routing
Choice Choice
Points to prioritized Route Groups
Route Route
Group Group
Route Group 1st 2nd
Performs digit manipulation Choice Choice
Points to the actual devices
GK
IP WAN PSTN
Devices
Gateways (H.323, MGCP)
Gatekeeper
Inter-cluster Trunk (remote CM)
Presentation_ID 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 56 V
Understanding Route Plans
User Dials Number
Route Pattern
The Route Pattern would represent an E.164 address
or address range that had some special route handling
Route Pattern needs. The Route Pattern points to a single Route List
Route List to perform this route handling. The Route Pattern
The Route List points to one or more Route Groups performs the digit manipulation (subtract or add digits)
that allow for Call Routing based on preference. This for the matched pattern. There is only one digit
is often looked at like a trunk group. The Route List manipulation table for any given Route Pattern.
can direct all calls to the Primary Route Group and
then use the Secondary Route Group if the Primary Route Group
is unavailable. Route List The Route Group points to one or more Devices
that allow for Call Routing based on preference. This is
often looked at like a trunk group. The Route Group can
direct all calls to the Primary Device and then use the
1st 2nd Secondary Device if the Primary is unavailable.
Choice Choice
Route Group Route Group
1st 2nd 1st 2nd

Choice Choice Choice Choice
Device
Typically a GW Device Device Device Device
or H.323 (GW) (GW) (GW) (GW)
Endpoint
(Netmeeting)
Internal vs. External Numbers
DN Route Pattern
Line #s 4005 9.1866XXXXXXX
GRP Pick 3510 9.@

Park 200 9.1[2-9]XX[2-9]XXXXXX
MEET_ME_CONFERENCE 361X 9.1XXXXXXXXXX
9.11
9.91!
Dialing Behavior
1000 Dialed Digits Cisco CallManager

<none> List
ListPotential
PotentialMatches
Matches
1 List
ListPotential
PotentialMatches
Matches
0 List
ListPotential
PotentialMatches
Matches
0 List
ListPotential
PotentialMatches
Matches
1 List
ListCurrent
CurrentMatch
Match
Call Setup
1001
Presentation_ID 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 59 IV
Digit Collection
Users dial string: Route Pattern

918
<off hook>>
91
9
9186
918665551212
91866555121
9186655512
918665551
91866555
9186655
918665
91866 9.1866XXXXXXXX
9.1866XXXXXXX Might Match
Match!
9.011! No Match
Might Match
9.1612[2-9]XXXXXX No Match
Might Match
CallManager actions:
9.[2-9]XXXXXX No Match
Might Match
Wait
Provide
Break
No otherdial
dial tone
patterns
tone 9.11 No Match
Might Match
Wait match
could
9.911 No Match
Might Match
Extend call
Inter-Digit Timeout
Users dial string: Route Pattern

911 <timeout>
9111
911<timeout><timeout>
<off <timeout>
hook> 9.1866XXXXXXXX
9.1866XXXXXXX Might
No Match
Match
9.@ Might
No Match
Match
Matches 1 billion+ digit 9.1[2-9]XX[2-9]XXXXXX Might

No Match
Match
CallManager actions:
strings 9.1XXXXXXXXXX Might
No Match
Match
Provide
Break
CallManager
Wait dial dial
tone
tone
will 9.11 Might
Match!
Match
Matches 10 for
wait million
the digit
stringsWait
timeout
9.91! Might
No Match
Match
Matches 1 digitmatch
Closest string
rules apply
Route Plan Configuration Process
MGCP Based H.323 Based

11 Add Gateway Devices VG200 All IOS GWs
V Cat 6K GWs V Device Protocol
2600 H.225
3600 Remote CallManager
Device Protocol
V
Inter-cluster Trunk
Route
RouteGroup
Group11 Route
RouteGroup
Group22
22 Build Route Groups Remote
RemoteCallManager
CallManager Catalyst
Catalyst6K
6KDigital
Digital
from available Devices 2600 (MGCP)
2600 (MGCP) 2600 (MGCP) Digital
2600 (MGCP) Digital
3640
3640 VG200
VG200
Route
RouteList
List11 Route
RouteList
List22
Build Route Lists from Route
RouteGroup
Group11 Route
RouteGroup
Group22
33 Route
available Route Groups RouteGroup
Group22 Route
RouteGroup
Group33
Route
RouteGroup
Group33 Route
RouteGroup
Group11
44 Build a Route Pattern to available Route Lists or Gateway Devices
Route Group Configuration
Route/Hunt List Configuration
Route Pattern/Hunt Pilot Configuration
Configure
Configureaaroute
routepattern
patternand
andpoint
pointititto
toaagateway
gatewayor
orroute
routelist
list
Dial-Peer Configuration on H323 Gateway
Local dial peer

Dial-peer
Dial-peer voice
voice <name>
<name> pots
pots
Destination-pattern
Destination-pattern <digits
<digits or
or wildcards>
wildcards>
Port
Port <x/x/x>
<x/x/x>
Network dial peer

Dial-peer
Dial-peer voice
voice <name>
<name> voip
voip
Destination-pattern
Destination-pattern <digits
<digits or
or wildcards>
wildcards>
Session
Session target
target ipv4:<ip
ipv4:<ip address
address of
of target>
target>
Route Plan Report
Route Plan Report
The following patterns/directory numbers are displayed:

call park numbers
call pickup numbers
conference numbers
route patterns
translation patterns
voice mail pilot
etc.
Save report data into a .csv file that you can import to other applications
The .csv file contains more detailed information than the web page and also includes
IP phone extensions
8. Telephony Class
of Service
General Definition
Class of Class of Class of

Service 1 Service 2 Service 3
Lobby Employee Executive
V IP WAN
Long Distance PSTN
International
Employee Executive
Partitions/ Calling Search Spaces:
Analogy with Subnets/ Access Lists
Access
Access List/
List/
Calling
Calling Search Space
Search Space Subnet/ Partition
permit
permit BB B
permit
permit CC
(implicit)
(implicit) deny
deny DD
Subnet/ Partition
Subnet/ Partition C
A
Subnet/ Partition
D
Partition Calling Search Space

Where you are Where you may call
Collects devices with similar Set of rules to set call
reachability characteristics restrictions/ permissions
Items placed in partitions: Directory Defines which partitions a
Numbers (DN), Route Patterns, Voice
Mail Ports... device may search to reach a
dialed number
Is assigned to IP phones, GWs
Partitions and Calling Search Spaces
Analogy
Rita wants to call Dave SWB Dallas Yellow Pages

Dave 972 813 5000
To do so, she needs to
know Daves number
Dave lists his number

In a directory
Dave
Rita
972 813 5000
Presentation_ID 2006 Cisco Systems, Inc. All rights reserved. Cisco Confidential 72 II
Analogy
To look up numbers, Rita

looks through the SWB Dallas Yellow Pages
directories she owns Dave 972 813 5000
If she doesnt have

the right directory
Ritas list of directories

SWB Dallas White Pages
Outlook Address Book
Little Black Book
she cant place the call

Dave
Rita
972 813 5000
Analogy
But if she has the SWB Dallas Yellow Pages

directory Dave has Dave 972 813 5000
listed his number in
Ritas list of directories

SWB Dallas Yellow Pages
Little Black Book
the call will go through

Dave
Rita
972 813 5000
Analogy
The list of directories in SWB Dallas Yellow Pages

which Rita looks up numbers
Dave 972 813 5000
is her calling search space
Ritas list of directories The directory in which

Daves number is listed
is his numbers partition
SWB Dallas Yellow Pages
Little Black Book
Dave
Rita
972 813 5000
Partition Definition
Partition Partition Partition Partition
Lobby Employee Executive Gateway
LobbyPT EmployeePT ExecutivePT Local and WAN

GatewayPT
Directory Numbers Directory Numbers Directory Numbers Route Pattern

63500 64050 64020 8.7XXXX (WANGWPT)
63501 64051 64021 9.1! (LongDisGWPT)
63502 64052 64022 9.011! (InternationalGWPT)
63503 64053 64023
A logical grouping of patterns.

All patterns in a partition are equally reachable.
Assigned to Directory Numbers and Route Patterns
Partition Configuration
Assigned
Assignedto
toDirectory
DirectoryNumbers,
Numbers,Route
RoutePatterns,
Patterns,Translation
TranslationPatterns
Patterns
Calling Search Space Definition
Lobby Phone Employee Phone Executive Phone Local and WAN

Gateway
Calling Search Space Calling Search Space Calling Search Space Calling Search Space
EmployeePT EmployeePT EmployeePT EmployeePT
WANGWPT ExecutivePT
LongDisGWPT WANGWPT
LongDisGWPT
InternationalGWPT
An ordered list of partitions.

Digit analysis looks through the callers calling search space (list of
partitions) when searching for the closest match for the callers dialed
number.
Assigned to Devices, Phones and Gateways, and Translation Patterns.
Calling Search Space Configuration
Assigned
Assignedto
toDevices
Devices(Phones,
(Phones,Gateways,
Gateways,etc.)
etc.)and
andTranslation
TranslationPatterns
Patterns
8. Media Resources
Media Resources
4 Resources are supported and shared:

Unicast conferecing
Media termination point
Transcoding
Music on hold
Why share resources?

Allow hardware and software resources to coexist.
Enable Cisco CallManager access resources available in the cluster.
Load distribution within a group of similar resources.
Allocate resources based on user preferences.
Conference Bridge (CFB)
In an Ad-Hoc conference, a conference controller can add participants to a

conference.
In a Meet-Me conference, the conference controller provides a bridge or directory
number for participants to dial.
Software and Hardware conference resources are available.
Unicast Conferencing
Phone A
Phone A
RTP
RTP CFB
RTP
RTP Phone C
Phone B Phone B
Conference Limits
Software Conference Limits

Up to 128 full-duplex streams are configurable.
48 users in a single conference or 16 conferencing resources with three
users per conference
Hardware Conference
Catalyst 6000Mixes all conference participants
G.711 or G.723 or G.729a or GSM/FR*
32 conference participants per physical port.
GSM/EFR
24 conference participants per physical port
Catalyst 4000 (G.711 only)Mixes 3 loudest talkers
24 conference participants per physical port
Maximum of 4 conferences with 6 participants in each.
Activate Cisco IP Voice Media Streaming App
Hardware Conference Bridge Configuration
Catalyst
Catalyst6000
6000port
portMAC
MACaddress
addressneeded
needed
Software Conference Bridge Configuration
Meet-Me Directory Number Configuration
Ensure
Ensuredirectory
directorynumber
numberor
orrange
rangeis
isunique
uniquewithin
withinthe
thecluster
cluster
Media Termination Point (MTP)
Cisco CallManager
Cisco IP Voice Media Streaming Application
SW MTP IP WAN
1002
PSTN
Router/GW
1001
H.323v1
Enables supplementary services to calls routed through an H.323v1 gateway.
Supplementary services are such features as:
call hold
call transfer
call park Incoming Stream
conferencing Initial Stream
Supplementary
Service Stream
Transcoder (XCODE)
XCODE Voice Messaging
G.729
G.729 G.711
G.711
MTP
Input
InputStream
Stream Stream
Streamfor
for
Supplementary
Supplementary
H.323 v1 Services
Services
MTP and Transcoding Limits
Media Termination Point

Up to 128 full-duplex streams are configurable.
64 resources are available for media termination point application.
Transcoder
Catalyst 6000
G723.1 to G.711, G.729a to G.711, GSM/FR to G.711, GSM/EFR to G.711
24 transcoders per physical port
G.729a to G.723.1, GSM/FR to G.729, GSM/EFR to G.723.1 or G.729a
Catalyst 4000
G729a to G.711
Transcoder Configuration
Catalyst
Catalyst6000
6000port
portMAC
MACaddress
addressneeded
needed
Music On Hold (MOH)
Types of Hold Cisco CallManager
User Hold Cisco IP Voice Media Streaming Application
Network Hold
Transfer hold
Conference hold
Call park hold
Audio Sources
Recorded Audio
Live Audio
Music On Hold
New Files are MOH Master Default MOH

automatically detected Audio Translator A
Service Storage Directory TFTP Server
and processed.
Cisco CallManager
Administration Copies
Audio Source files TFTPPath
Audio Source Input Directory
when they are mapped Directory
C:\Cisco\DropMOHAudioSourceFilesHere
Start B
Administrator copies Hard Coded MOH
MOH C Server Audio
audio source into Server
this directory Source Directory
H G D
E
I Kernel Mode RTP F DirectShow
Streaming Driver Filters
Music On Hold Server Limits
Up to 500 Unicast output streams of audio and 204 Multicast streams
simultaneously.
Up to 51 different audio sources can be configured.
Audio Translator Configuration
MOH server Configuration
MOH Service Parameters
Adding and Configuring Audio Source Files
10. Other
CallManager
Features
Call Park Configuration
Ensure
Ensurethat
thatCall
CallPark
ParkNumber/Range
Number/Rangeis isunique
uniquewithin
withinthe
thecluster
cluster
AND
AND
Each
Each Cisco CallManager that devicesare
Cisco CallManager that devices areregistered
registeredto
toneed
needtheir
their
own
ownunique
uniqueCall
CallPark
ParkNumber/Range
Number/Range
Call Park
Cisco 4 Dial 1234 to

CallManager pick up call
3 C
Sends call park code 5
to display on phone
1
1234
1 Initial Stream A B
Call Park Code 2 Call Park
5 Final Stream
Call Pickup/Group Call Pickup
Call Pickup
Allows a user to answer a call that is ringing on any phone in their Call Pickup Group
Group Call Pickup
Allows a user to answer a call that is ringing on any phone, if they know the Call Pickup
Group associated with that call
Call Pickup Configuration:
Step 1 of 2
Ensure
Ensurethat
thatDirectory
DirectoryNumber
Numberis
isunique
uniquewithin
withinthe
thecluster
cluster
Call Pickup Configuration:
Step 2 of 2
At
Atthe
theDirectory
DirectoryNumber
NumberConfiguration
Configurationof
ofPhone,
Phone,assign
assignCall
CallPickup
PickupGroup
Group
Call Pickup
Group Number 3500
In the same group or

DN=1111 not in the same group
1
Call Ringing
on 1111
DN=1112
3
4
Initial call
Call Redirected
DN=1113 RTP Stream

2 Press call
pickup soft key
Group Call Pickup
Group Number = 3500 Group Number = 3501
Call ringing
on 1112
1 4
DN=1111 DN=1112 DN=4113 DN=2111
DN=1113 DN=2112
2
Press group pickup soft
Initial call key and enters group
number 3500
Call Redirected
RTP Stream
Cisco IP Phone Services
Provide a dynamic and interactive environment LCD Display
among users, the enterprise, and the
Internetall through the Cisco IP Phone User
Interface
Utilize modern web technologies for application
services
XML-based data tags for phone content
processing
HTTP and TCP/IP for transport
Web servers and web scripting languages for
applications development
Softkeys Services
Button
Keypad Rocker
Key
IP Phone Services Configuration
Use
UseIP
IPAddress
Addressififyou
youdont
dontuse
useDNS
DNS
IP Phone Services Subscription
Service Subscription
Cisco IP Phone Services Examples
Meeting Room
Scheduler
World
Clock
Weather
Yellow Pages
Lookup
Flight Status
Stock Tracker
Transit
Schedules
Services Phone Display Examples
Menu Text Input
Image Directory Graphical

What is CallBack?
Phone A calls Phone B in the same cluster.

If Phone B is busy or no reply from Phone B then Phone A activates the
CallBack feature through CallBack soft key.
Now when Phone B becomes available, Phone A will receive an audio alert
and visual notification.
This operation is supported when

1. Busy or Call Forward Busy from called party; or
2. While no answer from called party, i.e.
Call forward no answer to Voice mail or any extension.
Call waiting on and ringing forever or just ringing.
Calling phone should be 7960 or 7940 that support soft keys

Supported phones
Call Back is supported as softkey on 7940, 7960 and

7970 etc. phones in 3.3 release
If called device is one of

30 SP+, 12 SP+,12 SP,12 S,30 VIP,IP Phone 7910 ,IP
Phone 7960, Phone 7935, Cisco VGC Phone, SCCP
Phone, CTI route point forwarding calls to above
devices.
with in cluster then CallBack feature supported.
How to configure CallBack(1)
Select SoftKey
template
Select Standard User

template
Press Copy
Standard User
template
Press insert the

template as Standard
User-CallBack
Click Configure soft key layout
Select On Hook and move CallBack

from left to right box by pressing right
arrow then press update button
Select Ring Out and move

CallBack from left to right box by
pressing right arrow.Order as first
key using up arrow.Press update
Select Device Pool to set this to all users
Select Standard User

CallBack and set this
template to Default (or set this
template to customized device
pool),if you want to enable
CallBack for all users .Press
update
How to activate CallBack
Check Cisco Extended

Functions and Update
11. Call Admission
Control (CAC) and
Automated Alternate
Routing (AAR)
Why Call Admission Control?
Example:
WAN bandwidth can only support two calls.
What happens when the third call is attempted?
Call #1
Call #2 X
IP WAN X
Call #3
X
Call #3
Causes poor quality for ALL calls
Many
Manytools
toolsto
togive
givevoice
voicepriority
priorityover
overdata.
data.
Call admission control is about preventing voice oversubscription.
Call admission control is about preventing voice oversubscription.
Bandwidth used for CAC
Codec Type G.711 G.723 G.729
Payload 64Kbps 8Kbps 8Kbps
CallManager Locations 80Kbps 24Kbps 24Kbps
Gatekeeper 128Kbps 20Kbps 20Kbps
IOS H.323 Gateways 128Kbps 16Kbps 16Kbps
Centralized Call Processing:
Locations based CAC
Location
Location11
Applications
(VMail, IVR, ICD, ...) SRST-enabled
PSTN router
CallManager
Cluster
Branch A
IP WAN
Headquarters
Branch B
Location
Location22
Location Configuration
Location
Locationis
isthen
thenassigned
assignedto
todevices
devices
AAR
Automated Alternate
Routing
Objectives
Introduce the AAR For Insufficient Bandwidth

feature
Learn how to configure AAR For Insufficient
Bandwidth feature.
Background
With Centralized Call Processing, call from one location to
another location without sufficient bandwidth
Phone A in
Dallas with
extension 1000
dials Phone B RTP
in San Jose at
extension 2000,
but no
bandwidth is
available.
Without AAR,
caller will get
reorder tone San Jose
and Not WAN Busy
Enough BW
display
Dallas
Function of AAR
The AAR For Insufficient Bandwidth feature provides a

mechanism to automatically reroute the call through
PSTN or other network by using an external/alternate
number when the call is blocked by Call Manager due
to insufficient location bandwidth, such that the caller
does not need to hang up and redial the called party
again.
In short, AAR is PSTN Backup for Locations.
AAR Service Parameters
The AAR feature is enabled by a cluster

wide CallManager service parameter
Automated Alternate Routing Enabled
AAR Configuration Service Parameters
AAR Configuration Service Parameters
How does AAR work?
With Centralized Call Processing, call from one location to
another location without sufficient bandwidth
Phone A in
Dallas with
extension 1000
dials Phone B RTP
in San Jose at
extension 2000,
but no
bandwidth is
available.
With AAR
enabled,
San Jose
WAN Busy
Dallas
12. Survivable Remote
Site Telephony (SRST)
What is SRST ?
Survivable Remote Site Telephony (SRST)
Capability in branch office routers for IP Telephony redundancy
Always available branch IP telephony (including calls from & to PSTN)
Ideal For Centralized CallManager Deployment
Licensed on number of users at remote site on IOS PLUS software
Survivability scales up to 480 users dependent upon platform
What is SRST ?
Central Call
Hub Site A Manager cluster
Survivable / Standby Remote Site
Telephony CCM CCM
Voice
Mail
Unique, industry-first capability in
branch office routers for IP CCM
Telephony redundancy
Always available branch IP primary
telephony
secondary
Ideal for enterprise customers
utilizing Centralized Cisco tertiary
CallManager deployment for IP
telephony at their branch offices
IP WAN
PSTN V
Spoke Site B
How It Works: Normal Operation
Central Site Call
Manager (CCM)
PSTN
Real Time
Protocol
Branch Router WAN
1750, 2600, 3600, Headquarters
7200, Cat4224
IAD2400
Keepalive:Ensures communication link with CallManager

Skinny: Protocol for call set up and tear down
SRST Configuration: Four Commands
SRST(config)#
call-manager-fallback
Enables Cisco CallManager fallback capability and puts you in a sub-

menu
SRST(config-cm-fallback)#
ip source-address <ip address> [port <port #>]
Enables router to receive skinny messages on this particular port. The

default port is 2000.
max-ephones
Defaults to 0. Maximum phones that will be allowed to register.

max-dn
Defaults to 0. Maximum number of directory numbers that can be

configured.
13. Extension Mobility
What is Extension Mobility?
Log onto any 7940/7960 (not 7910)
in a CM cluster to get extension
Device Profile includes: Extension,
Users office services, class of service restrictions
Single IP Phone 7960 applied to phone
(x5000) Login modes
Cluster
auto-logout other phones
LDAP keep login on other phones
Directory
IP Phone IP LAN Logout modes
Services explicit logout at phone
CRA Server
timed logout
7960
User logged onto phone
(Device Profile with x5000)
Example
ADP000011112222 Joes Profile
7000
1111
7001
Auto Generate Logout Login

Device Profile
SEP000011112222
Joe logs into device SEP000011112222
When a login is executed, the current configuration of a device is replaced by a
particular User Device Profile
When a logout is executed, the current configuration of a device (the User Device
Profile) is replaced by the Default Device Profile
Device Profile
A Device Profile is all the information which defines the configuration

of a particular device
Line Appearances, Speed Dials, Softkey templates, Services
Auto-Generated Device Profile
Represents a snapshot of an existing devices
configuration
Can only be used as a Default Device Profile for the
phone in logout state
Cannot be associated with a user
User Device Profile
Generated in the same manner as a device

Can be associated with a user in the same manner as a
device
Configurable by the administrator and associated to
users in the same manner as a device
What does Login Do?
When a login is executed, the current configuration of a

device is replaced by a particular User Device Profile
When a logout is executed, the current configuration of a
device (the User Device Profile) is replaced by the
Default Device Profile of the phone
Login interactions
The user selects the EM
Service on the phone
Login interactions
The user is prompted to enter
his/her user id and password
Login interactions
If users has more than one
device profile associated with
them, they are prompted to
select the profile they want to
log in with
Login interactions
The phone resets and the
users profile is loaded on the
phone
Logout interaction
On selecting the EM Service,
the user is shown a
confirmation screen
Configuration
Configuring the CallManager
Adding the EM Service
Adding the Service Parameters
Creating default device profile for users
Associating the user with these default device profiles
Configuring the Cisco IP Phones for Extension Mobility
Adding the EM Service
Go to the Cisco The HTTP trigger you created
CallManager
Administration pages
Select Feature > Cisco
IP Phone Services
Enter the information as
shown alongside
Always use IP
Address Always use device=#DEVICENAME#
Setting the services parameters
On the Cisco CallManager
Admin pages go to Service >
Service Parameters (machine
name) > Cisco Extension
Mobility
Enter the information as shown
alongside
Creating the device profile for a user
A User Device Profile is created in a

similar manner to the manual
creation of physical devices
Start with the Device Profile Menu
item
Adding a new user device profile
Device Profile Name and Phone Button Template are required
Configuring lines for the profile
Lines, Speed Dials, and Services may be configured after creation

Click Update Services and subscribe to EM service
Associate User Device Profile to User
User Device Profiles are

associated with Users by
selecting the Extension Mobility
link on the user pages
One or more device profiles can
be associated with the user
One of them could also be
selected as a default profile
Configuring the phones for extension mobility
The extension mobility configuration

is under the device configuration
page
Click on Subscribe/Unsubscribe
Services and subscribe the phone to
EM service
Enabling Extension Mobility for a phone
Enable Extension Mobility Feature must be checked

The items under this heading will show current device information
regarding the login status of the device
Assign Default Device Profile to a phone
Use Current Device Settings will create an Auto-Generated Device

Profile as the Default Device Profile
A User Device Profile may be assigned as the Default Device Profile by
selecting Select a User Device Profile
Click Update and reset device
Getting the Debug Traces
In the EM Service parameters page change the Debug traces on
parameter to true.
Restart the Cisco Tomcat service. NOTE: Restarting the Tomcat
service could disrupt other services running on Cisco Tomcat e.g.,
CAR, IPMA. The restart should be done at a non-critical time.
The debug traces are available under C:\Program
Files\Cisco\Trace\emapp
Also look at the Tomcat logs located at C:\Program
Files\Cisco\TomCat\logs\jasper
Errors and Remedies
Error Code Error message Remedy
0 Check if the CallManager services are

running
2,3 Check whether the user id/password is
configured in registry
HKEY_LOCAL_MACHINE\SOFTWARE\C
isco Systems, Inc.\Directory
Configuration\AppUsers\EMApp, if not then
the Directory install failed.
Device does In the phone configuration page, make sure

not allow Enable Extension Mobility Feature is
logon enabled and a logout profile is selected
Errors and Remedies
Error Code Error message Remedy
6,12 The user has no device profile associated

with him
9 Some problem with the LDAP directory
10 The EMApp user does not have

Authentication Proxy Rights enable. Go
to the DC Directory Admin and enable this
field for this user
Another user Logout the other user and try logging back
logged in
Errors and Remedies
Error Code Error Remedy
message
System not The EM Service is not enabled on the

enabled for server
login
User logged The system does not allow multiple

in elsewhere logins, either enable this feature on the
EM Service Parameters page or logout
user from other phone and ask him to
login back again
Errors and Remedies
Error Error Remedy

Code message
100 Check URL of the service, it

should end with
?device=#DEVICENAME#
101 Either the IIS on the EM Server

is down or the URL of the Login
Service is not configured in the
LDAP
14. Cisco IP
Communicator
Agenda
Product Overview
What is Cisco IP Communicator?
What is the difference between the Cisco IP Communicator and the
Cisco IP SoftPhone?
Software OverviewFeatures & Settings
Reference Material
For an excellent source of information, please see the Cisco IP
Communicator Administration Guide
Objectives
Understand the products features

Understand the difference between the Cisco IP Communicator
and the Cisco IP SoftPhone
Provide troubleshooting knowledge
Understand deployment options
Understand administration of the product
What is Cisco IP Communicator?
Turns your PC into an IP phone

Allows you to take your office phone
extension with you on the road
Allows teleworkers to access the
same phone services as working on
campus
Provides feature parity with Ciscos
advanced IP phones
Features & Benefits Overview
Feature parity with 7970 IP phones

Multiple lines
IP Phone services support (XML) like extension mobility
Integrates with CTI or IP Phone services application written by Cisco or
its partners
SCCP-based to work with Cisco CallManager 3.3(3) SR4
Call control features like shared lines, park, pickup, meet me conference,
etc.
Same manageability and scalability as Ciscos IP phones
Windows 2000 and Windows XP Professional support
Features & Benefits Overview
High-Quality Audio
Kernel-based media termination so that resource intensive applications will
not affect audio quality
Advanced jitter buffer
Speaker phone mode with echo suppression and noise cancellation
IP precedence marking (layer 3)
Audio Tuning Wizard for setting input and output levels
VPN support
Auto-detection of Cisco VPN client
Automated support for most VPN clients that look like NICs (e.g. Microsoft
PPTP client)
USB telephony device support
Removes need for keyboard emulation
Administration Overview
Unified administration with Cisco IP phones

Scaling 1 to 1 with IP phones (unlike TAPI-based)
Automatic software updates for deployment and management ease
Cisco CallManager redundancy support
Ability to manage user configurations
CDP support
Integrates with Cisco Emergency Responder
Whats not supported?
SRST Redundancy
Not supported in SRST V3.1 or earlier
CallManager Express
Not supported in CCME V3.1 or earlier
Localization
Support English & French only in IPC V1.1(4)
IPMA Manager
Extension Mobility
Not supported before CCM V3.3(4)
Video Telephony Advantage Application
15. Cisco VT Advantage
Cisco VT Advantage
Cisco IP Phone
7960G
Dialing
Association
Voice
Cisco VT Advantage
Answer
on PC with camera Video
PC associates with IP phone

Phone registers as a video capable phone
Convergence in
Initiate Voice/Video Call from IP phone Action!
Audio on the Phone
Video on the PC (Cisco VT Camera)
Features & Benefits
Interactive Face to Face Communications
Video calls just like regular phone calls
Video Codec & BW controlled by CM
H.263 & Cisco VT Camera Wideband codec support
Interoperable with 3rd party H.323 video terminals (Tandberg, NetMeeting)
Video Formats CIF
BW Override
Local & Remote Video check
Phone Configuration
QoS
Can be changed in
CCM Service
Parameter
Can configure DSCP
for Audio and Video
RTP port is hard
coded for 5445
16. Troubleshooting
Tools
Tools and Utilities to Configure,
Monitor and Troubleshoot CCM
CCM administration
Event log
CCM serviceability
Tools -> Real-Time Monitoring Tool
Trace configuration
SDL Traces
SDI Traces
Analysis (XML Files)
Q931 translator
Real-time Monitoring Tool (RMT)
From CallManager Administration, select the following:
Applications > Cisco CallManager Serviceability
Then from CallManager Serviceability, select the following:
Tools > Real-Time Monitoring Tool
RMT uses Windows 2000 Server PerfMon to facilitate monitoring a variety

of system variables in real-time
It can be installed on any PC besides CallManager server
If all counters are zero the service is stopped
Heart beat should increase once a second
Real-time Monitoring Tool (RMT)
CTI Application/Device/Line Searching & Monitoring.

Device Model Search & Monitoring.
More efficient and accurate device status polling.
H.323 Gatekeeper Perfmon Counters.
RMT Summary
RMT Server > CPU & Memory
RMT CallProcess > Call Activity
RMT CallProcess > Gateway Activity
RMT Service > Directory Server
RMT Device > Device Summary
RMT CTI > CtiManager
RMT CTI > CTI Search
RMT Perfmon > Perfmon Monitoring
Trace Configuration: SDL Trace
SDL (Signal Distribution Trace) allows the development engineer

(DE) to know that the code is functioning properly or to find the cause
of an error; Provides a direct interface to trace and alarms
SDL Trace can be directed to local files, Windows 2000 Event Log
and CiscoWorks 2000
Alarms are used to inform the administrator of unexpected events
(unable to access a file, database, Winsock or unable to allocate
other operating system resources)
From CCMAdmin, Application > CallManager Serviceability > Trace > Configuration
> CCM IP address > Cisco CallManager
Click Here to
Go to SDL
Trace Screen
Trace Configuration: SDI Trace
SDI (System Diagnostic Interface) files provide the greatest level of

detail
IP address, time stamps, TCP handle or
the device name can be used to find the occurrence of the request or
the disposition
of that request
You can search for the device name (SEPxxxxxxxxxxxx for a phone)
or IP address; You should find two occurrences for every request
(the request and what ever happened
to the request)
Trace Utility - Node, Service Selection
Select the Service on

Which Trace Needs
to Be Enabled
Select the
Node
Trace Utility - Configuration
Updates All CallManagers

in This Cluster
With These Settings
Click the Update Button

to Save the Settings
Trace Utility - Configured Services
Cisco CallManager
Cisco CDR Insert
Cisco CTI Manager
Cisco CTL Provider
Cisco Database Layer Monitor
Cisco Extended Functions
Cisco Extension Mobility
Cisco IP Manager Assistant
Cisco IP Voice Media Streaming App
Cisco Messaging Interface (CMI)
Cisco MOH Audio Translator
Cisco RIS (Real-time Information Server) Data Collector
Cisco Telephony Call Dispatcher (TCD)
Cisco TFTP
Cisco WebDialer
Trace Utility - Debug Trace Level
Error Used for all traces generated in abnormal path; Minimum amount
of CPU Cycles; this is the default
Special Non-repetitive messages; ex. all system and device
initialization messages
State Transition Call processing events
Significant Media layer events
Arbitrary Used for debugging excluding keepalives
Detail Detailed debug information; not recommended on systems
during normal system operation; CPU intensive
Trace Utility - Trace Directories
Directory Path: C:\Program Files\Cisco\Trace

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Cisco Unified

Changes made to the database

Database Client Computer

Device with extension 1000

To 7,500 IP Phones To 15,000 IP Phones To 30,000 IP Phones

Publisher and Publisher and

A cluster of four Cisco CallManagers

H.245 Master/Slave Determination Ack

H.245 Terminal Capabilities Set Ack

H.245 Open Logical Channel

Connection Acknowledge Connection Acknowledge

Station Off Hook

Dial-peer voice 100 voip

Dial-peer voice 101 voip

MGCP is, in essence, a master/slave protocol, where the

A point to point connection is an association between two

200 OK (RTP Port Info)

In the upper right, select Add a New Gateway

ccm-manager switchback <mode>

Controls how the gateway attempts to return management to the

Mode Configuration Options:

More Switchback Options:

Required MGCP statements:

Required CCM-Manager statements:

For Redundant configurations:

Required Dial-Peers: (ONLY the following)

For MGCP managed endpoints (those with

Automatically download your MGCP gateway

Router(config)# ccm-manager config server 10.0.0.1

* But you still need to setup MGCP in the CallManager first.

Users required to dial

CallManager routes two basic call types:

Route Group Route Group

1st 2nd 1st 2nd

GRP Pick 3510 9.@

1000 Dialed Digits Cisco CallManager

Users dial string: Route Pattern

Users dial string: Route Pattern

Matches 1 billion+ digit 9.1[2-9]XX[2-9]XXXXXX Might

MGCP Based H.323 Based

44 Build a Route Pattern to available Route Lists or Gateway Devices

Local dial peer

Network dial peer

The following patterns/directory numbers are displayed:

Class of Class of Class of

Lobby Employee Executive

Long Distance PSTN

Partition Calling Search Space

Rita wants to call Dave SWB Dallas Yellow Pages

Dave lists his number

To look up numbers, Rita

If she doesnt have

Ritas list of directories

she cant place the call

But if she has the SWB Dallas Yellow Pages

Ritas list of directories

the call will go through

The list of directories in SWB Dallas Yellow Pages

Ritas list of directories The directory in which

LobbyPT EmployeePT ExecutivePT Local and WAN

Directory Numbers Directory Numbers Directory Numbers Route Pattern

A logical grouping of patterns.

Lobby Phone Employee Phone Executive Phone Local and WAN