Transcript TECVVT-1001 Unified Communications Core

Unified Communications
Design and Deployment
Presentation_ID
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1
Agenda
Reference UC Architecture
Call Processing
Endpoints and Basic Call
Gateways
Media Resources
Call Admission Control
Dialplan
Features
TECVVT-1001
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Reference UC Architecture
Remote Sites
M
-30 Users Max
per site
Regional Office
-700 Users
M
M
M
MPLS
M
DS3
DS3
M
M
M
M
Headquarters
-3000 Users
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Network Infrastructure Recommendations
for Unified Communications
Unified CM
Cluster
PSTN
SRST
Router
IP WAN
Bandwidth
Provisioning
Campus
Campus
Access
Inline power
Multiple Qs
802.1p/Q
Fast link
Convergence
TECVVT-1001
Campus
Distribution
WAN
Aggregation
Multiple Qs
802.1p/Q
Classification
Reclassification
Multiple Qs
802.1p/Q
Traffic Shaping
Link Efficiency
(LFI, cRTP)
Classification
Reclassification
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Branch Office
Branch Router
Multiple Qs
802.1p/Q
Link Efficiency
(LFI, cRTP)
Classification
Reclassification
Branch Switch
Inline Power
Multiple Qs
802.1p/Q
4
Where To Get More Information
Network Infrastructure
Chapter
Cisco Unified Communications SRND
Based on Cisco Unified Communications
Manager
(One exists for each major release 4.x, 5.x, 6.x, etc…)
Where is the SRND?
http://www.cisco.com/go/designzone 
Design Zone for Unified Communications
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Agenda
Reference UC Architecture
Call Processing
Endpoints and Basic Call
Gateways
Media Resources
Call Admission Control
Dialplan
Features
TECVVT-1001
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Call Processing
Overview
Call Processing is…
 The sequence of operations performed by a switching
system from the acceptance of an incoming call through
the final disposition of the call.
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Call Processing
Cisco Call Processing Entities for the Enterprise
 Large Business
• Unified Communications Manager (Unified CM)
 Up to 60,000 phones per cluster
 > 30K, distributed architecture can include multiple clusters
 Medium-Small Business
• Unified Communications Manager Business Edition
(Unified CMBE)
 Up to 575 phones per server
 Server platform, No Clustering
 Co-Res with Unity Connection option
• Unified Communications Manager Express
(Unified CME)
 Up to 240 phones per server
 IOS Router platform, No Clustering
 Co-Res with Cisco Unity Express option
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Call Processing
Media Convergence Servers
+
=
 Unified CM is installed on a Cisco Media Convergence
Server (MCS) or VMWare ESX on UCS
 Unified CM 4.x: Cisco Windows OS (two CDs) + Unified
CM application
 Unified CM 5.x/6.x/7.x/8.x: Cisco appliance-based OS
and Unified CM application (one DVD)
 First server in cluster must be the Publisher server
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Call Processing
Unified CM Clustering: DB Replication and ICCS
Unified CM Cluster
Publisher
Database (DB)
Replication
DB
DB
CTI Manager
DB
ccm.exe
DB
ccm.exe
DB
DB
ICCS
ccm.exe
MoH Server
DB
TFTP Server
DB
ccm.exe
DB
Software Conferencing
MS-SQL/IDS Subscribers (Max. 19)
Unified CM 4.x:
Call Processing Servers (Max. 8)
DB=MS-SQL
| OS=MS W2K Server
Unified CM 5.x+: DB=IBM-IDS | OS=Linux
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Call Processing
Unified CM Clustering: Design Considerations
 The cluster appears as one entity, with a
single point of administration (the publisher)
 Several functions can be collocated on the same
server, depending on cluster size and server type
 Maximum of 19 subscribers per cluster (20 servers in
a cluster including the publisher)
 Maximum of eight call processing servers per cluster
 Maximum of 7500 IP Phones per Cisco Unified CM
server (server platform dependant)
 Maximum of 60,000 IP Phones per Cisco Unified CM
cluster (server platform and configuration dependant)
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Call Processing
Unified CM Clustering Capacities
 Unified CM Approximate Sizing
Server Platform
MCS 7845
Maximum Users
per Server
7500
High Availability
Server
Yes
MCS 7835
2500
Yes
MCS 7825
1000
No
 Cisco Unified Communications Sizing Tool
http://tools.cisco.com/cucst
• Accessible to Cisco and Partners only
• Good for you to know about!
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Unified CM Call Processing
Deployment Models
‘Call Processing’-Based Deployment Models
are dictated by:
 Physical Location of Unified CM cluster Servers
 Physical Location of Unified CM cluster IP Phones
 Number of Unified CM clusters
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Unified CM Deployment Models
Single Site
 Unified CM,
applications and DSP
resources at same
physical location
Applications
(VMail, IPCC, MP…)
 Supports up to
30,000 SIP or SCCP
phones per cluster
Unified CM Cluster
 PSTN used for
all external calls
PSTN
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Deployment Models
Centralized Call Processing
Applications
(VMail, IPCC, MP…)
PSTN
SRST-Enabled
Routers
Unified CM
Cluster
IP WAN
Branch A
Headquarters
 Unified CM cluster at central/HQ site
 Applications and DSP resources can be centralized or distributed
Branch B
 Supports up to 60,000 SIP or SCCP phones per cluster
 If WAN is “busy”, transparent use of PSTN (Automated Alternate Routing—AAR)
 Survivable Remote Site Telephony (SRST) for remote branches
 Maximum 1000 sites per cluster (500 prior to Unified CM 6.x)
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Deployment Models
Distributed Call Processing (Unified CM-Unified CM Model)
Applications
(VMail, IPCC, MP…)
Applications
PSTN
Unified CM
Cluster
GK
Unified CM
Cluster
IP WAN
Regional Branch A
Headquarters
GK
Gatekeeper
 Unified CM, applications, and DSPs located at
each site
 Up to 60,000 SIP or SCCP phones per cluster
Applications
Unified CM
Cluster
 ~100 sites
 Transparent use of PSTN if IP WAN unavailable
 Each cluster can be single site or centralized call
processing topology
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Regional Branch B
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Deployment Models
Clustering over the WAN (CoW)
Unified CM Cluster
Voice Mail
San Jose
Voice Mail
Distance
San Francisco
 Unified CM servers in a cluster separated by WAN for spatial redundancy
 Applications may be located at each site, thus separated by WAN
 Single point of administration, feature transparency (e.g. Extension Mobility),
unified dial plan
Increased to
80-ms RTT
in 6.1
 Maximum 40-ms round-trip delay between any two Unified CM across the WAN
 900 kbps bandwidth for each 10,000 BHCA between sites
 Maximum of eight active locations
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B/W Required
Increased in
6.1
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Unified CM Call Processing
Failover and Redundancy
Gateways
Unified CM Subscriber
DSP Resources
Conferencing
DSP Resources
Transcoding
Conf
Xcode
Intra-Cluster
Communications
(ICCS)
Cisco Unity
Vmail Server
Unified CM Subscriber
JTAPI
IP-IVR
Active
Unified CM Server
IP Phones
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Directory Services
Music on Hold
Software Conferencing
Software MTP
TFTP
Call Processing
CTI/QBE I/F
SCCP I/F
MGCP I/F
H.323 I/F
SIP I/F
Directory Services
Music on Hold
Software Conferencing
Software MTP
TFTP
Call Processing
CTI/QBE I/F
SCCP I/F
MGCP I/F
H.323 I/F
SIP I/F
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Unified CM Redundancy
1:1 vs. 2:1 Redundancy
MCS 7835 Supports 2500 Phones/Server
2:1 Redundancy Scheme
Backup
1 to
2500
2501 to
5000
1:1 Redundancy Scheme
1 to
1250
1251 to
2500
2501 to
3750
3751 to
5000
 Cost-efficient redundancy
 Load-sharing redundancy
 High Availability
during upgrades
 High Availability
during upgrades
 Maximum of 10,000 backup
registrations/server
 Faster failover
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Unified Communications Infrastructure
Failover and Redundancy: 1:1 Redundancy Example
 MCS 7845 supports 7500
phones/server
Phone Set 1
To 7,500 IP Phones
Phone Set 2
To 15,000 IP Phones
Publisher and
TFTP Server(s)
1 to 3750: Primary
3751 to 7500: Backup
3751 to 7500: Primary
1 to 3750: Backup
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 Load-share between primary
and backup servers
Publisher and
TFTP Server(s)
1–
3750
7501–
11,250
Cisco Public
3751–
7500
11,251–
15,000
To 30,000 IP Phones
Publisher and
TFTP Server(s)
1–
3750
3751 to
7500
7501–
11,250
11,251–
15,000
15,001–
18,250
18,251–
22,500
22,501–
26,250
26,251–
30,000
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Unified Communications Infrastructure
Failover and Redundancy: Survivable Remote Site Telephony
Normal
WAN Operation
Failure
Signaling Traffic
Unified CM
Cluster
Signaling Traffic
IP WAN
Branch Site
SRST
Capable
Router
Voice Traffic
PSTN
Applications
Central Site
Voice Traffic
 IP Phones have SRST router IP as the last option in their CM GROUP configuration
 Support for both SIP and SCCP IP Phones
 With SRST, only a subset of features are available to the phones (DID, DOD, call
hold, transfer, speed dial, caller ID, etc.)
 H323 PSTN GW connectivity option during failure modes via VoIP/POTS dialpeers; MGCP GWs require the ‘MGCP Fallback to H323’ feature
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Agenda
Reference UC Architecture
Call Processing
Endpoints and Basic Call
Gateways
Media Resources
Call Admission Control
Dialplan
Features
TECVVT-1001
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Unified Communications Core:
Endpoints and Basic Call
Topic Overview
 Endpoints
Not just phones—but all devices that terminate and originate
IP telephony sessions: Gateways, MCUs, etc., in addition to user phones
Come with various capabilities and support varied protocols
Phones interface directly with users and define User Experience
 Network Services
Provides functions other than Call Control such as: Inline Power, Device
Discovery and Authentication, VLAN settings, IP Addressing, and other
operating parameters
 Line Side and Trunk Side Protocols
Call control protocols and initiate, negotiate, and tear down calls
Line side protocols enable user facing services and have richer feature
support
Trunk side enables connectivity with other telephony systems and
application servers and interoperability is a prime criteria
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思科IP电话终端一览
9900 Series
9900 Series
8900 Series
8900 Series
Voice
7900 Series
SPA 500
Series
SPA 900
Series
SPA 9000
TECVVT-1001
SPA 500
Series
UC-500
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7900 Series
7900 Series
6900 Series
6900 Series
6900 Series
3900 Series
3900 Series
3900 Series
Unified
Unified CMBE
CME on ISR
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PORTFOLIO AVG LIST PRICE ($USD)
Voice/
Video/
Apps
IP PHONE APPLICATIONS
Higher
7900 Series
Lower
Unified CM
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Unified Communications Core:
Network Services
IP Phone Boot-Up Process
1. Inline Power (ILP)
Inline Power Initialization
2. Cisco Discovery Protocol (CDP) or Link Layer
Discovery Protocol-Media Endpoint Discovery
(LLDP-MED)
ILP Negotiation, Voice VLAN ID
3. Dynamic Host Configuration Protocol (DHCP)
IP Assignment, TFTP Server Allocation, DNS (optional)
4. Trivial File Transfer Protocol (TFTP)
Configuration File, IP Phone Firmware
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Unified Communications Core:
Network Services
DHCP
Inline Power Provided
Cisco
Catalyst
Switch
CDP/LLDP Neighbored
DHCP Req
DHCP Rsp (IP Add, Def-GW, TFTP, DNS*)
DHCP
Server
Option 150 or Option 66
 Phone displays: “Configuring IP”
(DNS is optional)
DHCP Request Must Be Made in
the Correct VLAN to Place the
Phone in the Correct Subnet!!
 Phone settings: Settings=>NetCfg=>“DHCP Server”
Settings=>NetCfg=>“IP Address”
Settings=>NetCfg=>“TFTP Server X”
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Unified Communications Core:
Network Services
TFTP
Unified CM
Cluster
UCM1
MAC Address:
001956A6A7ED
UCMx
UCM2
Backup Link
Publisher
TFTP
TFTP: GET Configuration File(s) for MAC
Phone Configuration, Firmware Download
(If Required)
1=UCM1: 10.1.1.1
2=UCM2: 10.1.1.2
…
CM Group: UCM1
UCM2
Device Pool
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Unified Communications Core: Basic Call
Centralized Deployment Model
IP Phone to IP Phone Example
 Call Processing is essentially the same in this
deployment model as in the single site case; IP
makes the technology more topology independent
Unified CM
IP Phone A
Media
IP WAN
Dial Plan Lookup
IP Phone B
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Unified Communications Core:
Network Services
Signaling Protocols
Gateways
Unified CM 5.x+
Unified CME
Conf/ Xcode
Carriers/
Other Vendors
PBXs
Unified CM 5.x+
DSP Resources
Cisco Unified
Presence Server
Rich-Media
Conferencing
Soft
Phones
Unified
Messaging
Microsoft
LCS/OCS
IBM
Sametime
CTI Apps
Cisco and
3rd-party Phones
Video
Endpoints
Cisco Unified
Personal
Communicator
Cisco Unified Communications provides seamless interworking with
SIP, H.323, MGCP, SCCP, TAPI/JTAPI and Q.SIG Protocols
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SCCP
MGCP
H.323
CTI
SIP/SIMPLE
CSTA over SIP
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Unified Communications Core:
Network Services
Signaling Protocols: Unified CM as “Protocol Translator”
Session Initiation Protocol
IP Phones
Skinny Client Control Protocol
ITU-T H.323 Standard
SIP Networks
IP Phones
Gateways
Analog
Phones
Wireless IP Phones
Gateways
SIP
Analog Phones
PC-Based
IP Phones
Video
Terminals
Computer Telephony Integration/
Quick Buffer Encoding
Media Gateway Control Protocol
Gateways
Applications
Servers
(JTAPI/CTI)
Analog
Phones
Call Agents
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Unified Communications Core:
Endpoints and Basic Call
Reference Requirements and Recommendations
 Endpoint Selection
Executive full featured phones; some with video
Conference rooms with high quality speaker phones
Deskphones with services
 Inline Power for Phones
Use Compatible access switches
 VLAN assignment
Keep the Voice and Data networks separated out
 DHCP Addressing
Use a central DHCP server and independent servers in larger branches;
keep address lease times large in branches using central server
 TFTP Server
Single TFTP Server co-resident with Publisher
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Unified Communications Core
Endpoints and Basic Call
Use Central DHCP
Server
Integrated DHCP
Server
MPLS
Publisher &
TFTP Server
Independent Central
DHCP Server
2nd TFTP Server
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Agenda
Reference UC Architecture
Call Processing
Endpoints and Basic Call
Gateways
Media Resources
Call Admission Control
Dialplan
Features
TECVVT-1001
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Unified Communications Core:
Gateways and CUBE
Topic Overview
 Gateways
Convert between IP and TDM Networks for PSTN access
Distinguished by capacity, technology, IP Telephony protocol, services
(fax, modem, video, etc.)
 Cisco Unified Border Element
Formerly known as Multi-service IP-IP Gateway
Used as a demarcation point between two IP Telephony networks—
such as between the enterprise and IP telephony provider
 Deployment Models
Centralized: All PSTN access through the main HQ
Distributed: Each site has local PSTN access; with both centralized
or distributed call control
Hybrid: A optimization of the above
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Unified Communications Core: Gateways
Gateway Selection Criteria
Cisco Unified CM
PSTN
Router/
Gateway
IP WAN
 Voice port density requirements
 Signaling protocol (H.323, MGCP, SIP, etc.)
 Support for required PSTN signaling types
 Support for required WAN interfaces and QoS
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Unified Communications Core: Gateways
H.323
TDM
IP
PSTN
PRI Layer 3
Layer 2
Framing
H.225
H.245
Cisco Unified CM
 All PSTN signaling terminates on gateway
 H.225 communication between gateway and Cisco
Unified CM
 H.323 is a “peer-to-peer” protocol: each side can
make decisions
TECVVT-1001
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Unified Communications Core: Gateways
The Power of Cisco IOS Dial-Peers: H.323 and SIP
PSTN
dp 10 pots
dp 1 voip
dp 11 pots
dp 2 voip
dp 12 pots
dp 3 voip
IP
Dial-Peers Allow You to:
 Switch calls intelligently if required (interpret the dial plan)
 Digit manipulation (called, calling and numbering plan)
 Failover (preferences) to alternate destinations
 Load balancing
 Video ISDN switching
 Insert applications into the call path: TCL/VXML
Build support for signaling variations (e.g. CLID on T1 CAS)
Hookflash trunk release on FXO
VXML call control for call centers
Redistribute calls-in-q for CVP
AA in the GW
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These Capabilities
Do Not Exist for
MGCP-Controlled GWs
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Unified Communications Core: Gateways
Protocol and Platform Recommendations
Designing UC Gateways and
DSP Engineering in Enterprise
Networks
Cisco
Unified CM
Cluster
PSTN
SRST
Router
BRKVVT-2010
Router/
Gateway
IP WAN
Central Site
Remote Site
 H.323, SIP, MGCP fallback
to H.323/SIP
 Standalone, Router-integrated
 Platforms:
 MGCP, SIP, H.323
 Standalone, Router-integrated
 Platforms:
WS-X6608, CMM
26XX, 28XX
37XX, 38XX
17XX, 18XX
26xx, 28XX
37xx, 38xx
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Cisco Unified Border Element
Key Features
CUBE
Session Mgmt
Demarcation
Real-time session Mgmt
Call Admissions Control
Ensuring QoS
PSTN GW Fallback
Statistics and Billing
Redundancy/Scalability
Fault isolation
Topology Hiding
Network Borders
L5/L7 Protocol Demarc
Statistics and Billing
Interworking
Yours
H.323 and SIP
SIP Normalization
DTMF Interworking
Transcoding
Codec Filtering
Fax/Modem Support
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Security
Encryption
Authentication
Registration
SIP Protection
FW Placement
Toll fraud
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Unified Communications Core:
Cisco Unified Border Element
Usage in IP-PSTN Connectivity
Cisco Unified Communications
Manager Cluster
Cisco Unified
Border Element
H.323/SIP
Trunk
SBC
IP-PSTN
Service
Provider
SIP
Trunk

Network/Topology Hiding for Voice and Video Calls

Signaling Interworking

Protocol Support—H.323 and SIP

Supplementary Services

Voice Codecs—G.711, G.729, G.726, G.723, G.728, Transparent

Transcoding

Video Codecs—H.261, H.263 and H.264

Transport Mode—TCP, UDP

Codec Filtering

Number Translation

Media—Media Flow Through and Media Flow Around

Quality of Service

DTMF Interworking—H.245 Alphanumeric, Signal, RFC2833,
SIP NOTIFY

Call Admission Control

Call Detail Records

Fax/Modem—T.38, Passthrough, Cisco Fax Relay,
Modem Passthrough

TCL/VXML Support

Security—TLS, IPSec with SRTP

Rotary Support
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Unified Communications Core:
Cisco Unified Border Element
BRKVVT-2305
Integrating Voice and Video
over IP Networks Using the
Cisco Unified Border Element
Address Hiding
192.168.10.10
192.168.10.50
10.10.10.5
10.10.10.6
CUBE
Cisco Unified
CM Cluster
192.168.10.10
CUBE
Site A
Unified Border
Element
Site A—192.168.10.x/24
192.168.10.50
IP WAN
IP WAN
10.10.10.x/24
Site B
Unified Border
Element
Cisco Unified
CM Cluster
Site B—192.168.10.x/24
 Within the same company—between departments
having overlapping addresses
 Integrating new acquisition into the existing
voice network
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Agenda
Reference UC Architecture
Call Processing
Endpoints and Basic Call
Gateways
Media Resources
Call Admission Control
Dialplan
Features
TECVVT-1001
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Unified Communications Core:
Media Resources
Topic Overview
 Role of Media Resources
Necessary where any manipulation of media is required such as mixing
(conference bridges), changing the compression type (transcoding), etc.
 Types of Resources
Hardware (DSP Based)
Software (IOS Based)
Software (IP Voice Media Streaming Application)
 Deployment Models
Centralized: Media flows to the central site over company WAN
increasing BW requirements put perhaps save in hardware costs
by aggregation
Distributed: Save in WAN BW by local media services but perhaps
at a higher cost
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Unified Communications Core:
Media Resources
Conferencing, Transcoding, Music on Hold
 Conference Bridge
IVR
Cisco
Unified
CM
Cluster
Music
on Hold
DSPs needed for multicodec conferences
 Media Termination Point
Transcoder
Xcode
Media Termination
MTP
MTP
DSPs optional
Conference
Bridge
Conf
 Transcoding
DSPs needed to transcode
multiple CODEC types (e.g.,
G.711 to G.729)
PSTN
IP WAN
Automatic codec selection
 Music on Hold
Multiple source types possible
(centralized or branch-based)
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Unified Communications Core:
Media Resources
Media Resource Group Lists and Media Resource Groups
User Needs
Media
Resource
Media
Resource
Manager
Assigned to Device
Directly or via
Device Pool
Media
Resource
Group List
1st
Choice
2nd
Choice
Media
Resource
Group
1st
Choice
Media
Resource
1
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Resource
Group
2nd
Choice
Media
Resource
2
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1st
Choice
Media
Resource
3
2nd
Choice
Media
Resource
4
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Unified Communications Core:
Media Resources
MRGL and Device Association
MRGL—A
MRG—A
Device Pool—A
 CM Group
 Date/Time Group
Media
Resources
 Region
 Media Resource Group List
Site A
MRGL—B
MRG—B
Media
Resources
Assign a MRGL Directly
to the Device Take a
Higher Priority than
Device-Pool Based
Configuration
Device Pool—B
 CM Group
 Date/Time Group
 Region
 Media Resource Group List
For Groups of Devices that Don’t
Need Special Media Resources or
Can’t Be Assigned a MRGL Directly
Assign the MRGL via the Device Pool
Site B
TECVVT-1001
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47
Unified Communications Core:
Media Resources
Centralized vs. Distributed DSPs
Cisco
Unified CM
Cluster
X
PSTN
Centralized
DSPs
A
IP WAN
B
Conf
Branch
Central
Site
$ Bandwidth vs. $ Hardware
X
Distributed
DSPs
A
PSTN
IP WAN
B
Branch
TECVVT-1001
Cisco
Unified CM
Cluster
© 2009 Cisco Systems, Inc. All rights reserved.
Conf
MRG
Cisco Public
Conf
Conf
Conf
Conf
MRG
MRG
Central
Site
48
Unified Communications Core:
Media Resources
Key Takeaways
 CFB, MTP, XCODE, MOH are media resources
 Media ResourceMRGMRGL
 Load Balance (round-robin) Similar Media Resources within
an MRG
 MRM walks through MRG in order top-down
 Next MRG in MRGL is used required resource is exhausted
or has failed (unregistered)
TECVVT-1001
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49
Unified Communications Core
Media Resources
dsp
dsp
dsp
MPLS
dsp
dsp
dsp
dsp
dsp
dsp
dsp
dsp
dsp
dsp
dsp
TECVVT-1001
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50
Agenda
Reference UC Architecture
Call Processing
Endpoints and Basic Call
Gateways
Media Resources
Call Admission Control
Dialplan
Features
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51
Call Admission Control
Why Is It Needed?
Circuit-Switched
Networks
Packet-Switched
Networks
IP WAN
PSTN
Physical
Trunks
IP WAN
Link
Third Call
Rejected
PBX
Router/
Gateway
STOP
IP WAN Link’s LLQ Is
Provisioned for Two
Calls (Equivalent to
Two “Virtual” Trunks)
No Physical Limitation on
IP Links
UC
Third Call Can Go Through,
Manager but Voice Quality of All
Calls Degrades
 Call Admission Control
Blocks Third Call
TECVVT-1001
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52
Unified Communications Core
Call Admission Control
Two Types of CAC:
BRKRST-2505
Call Admission Control
Design for Unified
Communications
BRKVVT-3303
Advanced Call Admission
Control Design,
Implementation, and
Troubleshooting Using
RSVP
A. Topology-Unaware
B. Topology-Aware
Three Options for Configuring CAC:
1. Unified CM Locations (Topology-Unaware)
2. RSVP (Topology-Aware)
3. Gatekeeper Zones (Topology-Unaware)
TECVVT-1001
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53
Unified CM Static Locations
Concept
Central
Site
 Prevent WAN link oversubscription by limiting
voice bandwidth
<Hub_None>
Location
 Assign bandwidth limit for
voice per location
(G729 = 24Kbps, G711=80Kbps)
 Location1 makes a G729 call
over WAN to the Central Site
IP WAN
PSTN
1
Location 1
Max BW = 24 kbps
Avail BW = 24 kbps
TECVVT-1001
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Remote
Sites
Location 2
Max BW = 48 kbps
Avail BW = 48 kbps
54
Unified CM Static Locations
Concept
Central
Site
 Prevent WAN link oversubscription by limiting
voice bandwidth
<Hub_None>
Location
 Assign bandwidth limit for
voice per location
(G729 = 24Kbps, G711=80Kbps)
 Location1 makes a G729 call
over WAN to the Central Site
 Location 1 attempts a
G729 call, but Locationsbased CAC blocks the call
 When resources are
insufficient, by default the user
hears a fast-busy tone and a
configurable message is
displayed
TECVVT-1001
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IP WAN
PSTN
2nd
2
STOP
1
Location 1
Max BW = 24 kbps
Avail BW = 0 kbps
Remote
Sites
Location 2
Max BW = 48 kbps
Avail BW = 48 kbps
55
Unified CM Static Locations
Concept
Central
Site
 Prevent WAN link oversubscription by limiting
voice bandwidth
<Hub_None>
Location
 Assign bandwidth limit for
voice per location
(G729 = 24Kbps, G711=80Kbps)
 Location1 makes a G729 call
over WAN to the Central Site
 Location 1 attempts a
G729 call, but Locationsbased CAC blocks the call
2nd
 When resources are
insufficient, by default the user
hears a fast-busy tone and a
configurable message is
displayed
TECVVT-1001
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3
2
IP WAN
PSTN
STOP
1
Location 1
Max BW = 24 kbps
Avail BW = 0 kbps
Remote
Sites
Location 2
Max BW = 48 kbps
Avail BW = 48 kbps
56
Unified CM Static Locations
Bandwidth Provisioning
Provision LLQ PQ
with These Values
Actual L3
Bandwidth
L2 Bandwidth
(Frame Relay)
80 Kbps
80 Kbps
(64K + Header)
81.6 Kbps
(80K + L2 Header)
24 Kbps
24 Kbps
(8K + Header)
25.6 Kbps
(24K + L2 Header)
CUCM Location
G.711 Audio
G.729 Audio
384K Video
TECVVT-1001
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384 Kbps
Cisco Public
420 Kbps
(384K + Est. L2/L3 Headers
57
Unified CM Static Locations
Notes
 Audio is represented as a static bit-rate + IP overhead (i.e.
24k for G.729, 80k for G.711)
 Video is represented as a static bit-rate only (i.e. 384k for a
384k call) which includes the audio portion
 The audio bandwidth setting does not pertain to the audio
channel of a video call
 If transcoders are needed (e.g., in presence of G.711-only
devices), the transcoder must be colocated with the G.711only device
 The location setting on CTI route points is only used by
Cisco Unified CM if an application registers to handle media
with that route point
TECVVT-1001
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58
Call Admission Control
Simple Hub and Spoke Topologies: Centralized Deployments
 Use Static Locations: one
location per spoke site
Hub Site
 Devices at hub site in
<Hub_None> location
 Up to 1,000 locations per
Cisco Unified CM cluster
Increased to
2000 in 7.1
 If more than one Cisco Unified
CM cluster at hub site, use
Intercluster Trunks (leave in
<Hub_None> location)
 Location needs to be updated if
device moves to a different site
Loc.
<Hub_None>
...
Loc. 1
Loc.1000
Spoke Sites
TECVVT-1001
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59
Agenda
Reference UC Architecture
Call Processing
Endpoints and Basic Call
Gateways
Media Resources
Call Admission Control
Dialplan
Features
TECVVT-1001
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60
Off-Cluster Routes in Unified CM
Overall Structure
Route Pattern
 Matches dialed number for external calls
 Points to a route list for routing
 Performs digit manipulation (optional)
Route
List
Route List
 Points to prioritized route groups
 Performs digit manipulation (opt)
Route Group
2nd Choice
1st
Choice
Route
Group 1
Route
Group 2
 Points to the actual devices
 Distribution algorithm
1st
Choice
IP WAN
Devices
 Gateways (MGCP, SCCP, H.323)
 Gatekeeper (H.323)
 Trunk (H.323, ICT, SIP)
TECVVT-1001
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GK
2nd
Choice
PSTN
Configuration Order
Route
Pattern
61
Cisco Unified CM Call Routing Logic
Commonly Used Wildcards
Delimiter (Does Not Match Any Digits)—Used for Discarding
Range of Digits (Between 2 and 9)
Single Digit Between 0 and 9
9 . [2-9] XXXXXX
One or More Occurrences of Digits Between 0 and 9
The “#” Digit—Used to Avoid InterDigit Timeout
9.011! #
A Macro That Enters the Whole North American
Numbering Plan into Cisco Unified CM
9. @
TECVVT-1001
(or a Different Country’s Numbering Plan If Using the
International Dial Plan Tool)
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62
Cisco Unified CM Call Routing Logic
Matching Patterns
1111
Matches 1111
*1*1
Matches *1*1
12XX
Matches Numbers Between 1200 and 1299
13[25-8]6
Matches 1326, 1356, 1366, 1376, 1386
13[^3-9]6
Matches 1306, 1316, 1326, 13*6, 13#6
13!#
TECVVT-1001
Matches Any Number That Begins with 13, Is
Followed by One or More Digits, and Ends with #;
135# and 13579# Are Example Matches
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63
Cisco Unified CM Call Routing Logic
Basic Principle
Cisco Unified CM Call Routing Logic
Route Patterns
User Dials
“1200”
1XXX
12XX
Directory Numbers
User Dials
“1234”
1234
1234
 Cisco Unified CM matches the most specific pattern
(longest-match logic)
 For call routing, an IP phone directory number acts
as a ‘route pattern’ that matches a single number
TECVVT-1001
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64
Building Classes of Service
Concepts
PartitionA
CSS1
CSS2
Lines
PartitionB
CSS3
Gateways
PartitionB
PartitionA
“Dialable” Patterns
“Dialing” Devices
Phones
2002
2001
2000
PartitionA
PartitionB
Applications
TECVVT-1001
PartitionA
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(Directory Numbers)
7 [Transform Mask: 2001]
Translation
Patterns
911
9.[2-9]XXXXXX
Route
Patterns
9.[2-9]XX[2-9]XXXXXX
PartitionB
5000
900X
99XX
CSS4
Lines
Application Numbers
(CTI Route Points, CTI Ports)
Special numbers
(MeetMe, CallPickup...)
8001
8000
Voice Mail Ports
9.011!
Route
Patterns
65
Dial Plan Example
Single Site Deployment Model: Composite View
Calling Search
Spaces
Calling
Search
Space
Assigned
Partitions
Internal
All IP Phone DNs
Route
Groups
Devices
Route
Patterns
911
9.911
Internal Only
Local
Local
Route
Lists
LOC
RL
LOC
RG
9.[2–9]XXXXXX
9.[2–9]XX[2–9]XXXXXX
National
PSTN
National
2
9.1[2–9]XX[2–9]XXXXXX
International
International
LD
RL
LD
RG
1
9.011!
9.011!#
TECVVT-1001
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66
Dial Plan Example
Centralized Deployment Model: Composite View
Calling
Search
Space
Assigned
Calling Search
Spaces
Route
Groups
Devices
NYC911
911
9.911
NYCInternal
NYC
Phones
Partitions
Route
Lists
NYCAllCalls
Route
Patterns
NYCPSTN
9.[2–9]XXXXXX
9.1[2–9]XX[2-9]XXXXXX
9.011!
9.011!#
NYC
PSTN
NYC
PSTN
PSTN
NYC
Gateways
Internal
All IP Phones
PHLInternal
PHL
Phones
TECVVT-1001
PHLAllCalls
© 2009 Cisco Systems, Inc. All rights reserved.
PHL911
911
9.911
PHLPSTN
9.[2–9]XXXXXX
9.1[2–9]XX[2–9]XXXXXX
9.011!
9.011!#
Cisco Public
PHL
PSTN
PHL
PSTN
PSTN
PHL
Gateways
67
Dial Plan Example
Centralized Deployment Model: Composite View
~ Using Local Route Group feature of Unified CM version 7.0 ~
Calling
Search
Space
Assigned
Calling Search
Spaces
NYCInternal
Partitions
Route
Lists
Route
Groups
Internal
All IP Phones
NYC
Phones
Devices
NYCAllCalls
PSTN
US EMERG
911
9.911
US
PSTN
Local
Route
Group
NYC
Gateways
US PSTN
PHLInternal
PHL
Phones
TECVVT-1001
PHLAllCalls
© 2009 Cisco Systems, Inc. All rights reserved.
9.[2–9]XXXXXX
9.1[2–9]XX[2-9]XXXXXX
9.011!
9.011!#
Route
Patterns
Cisco Public
PSTN
Route group chosen
as per device pool of
calling device
PHL
Gateways
68
Local Route Group
with it – key take aways
 We go from route patterns that are site-specific to
patterns that are type-specific.
e.g.: local, national, international
 We now group by dial plan domains
e.g.: US dialing habits of 9 plus seven, 9 plus ten, 91 plus ten,
9011 plus ???, 911, 9911). I could not add a French site to the
preceding example without creating patterns for 112, 0112, 00[16]XXXXXXXX, 000!, 000!#
 We get site-specific failover for “free” on long distance
patterns
 We now have much fewer things to configure per site
TECVVT-1001
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69
Unified Communications Core
Dial Plan
UC Reference Requirements and Recommendations
Design Criteria:
 Centralized Management
 “9” PSTN Access Code
 Three Classes of service
 Centralized IP PSTN for Long Distance
and International calling
Design Recommendations:
 Class of Service
 Route Groups/Route Lists
 Unified CM 7.x
TECVVT-1001
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70
Translation Patterns
Key Concept
 Looks like a route pattern, allows digit manipulation
 Instead of sending calls outside via a route list, forces
second lookup in Cisco Unified CM, using a (possibly
different) calling search space
User Dials
“0” to Reach
Operator
TECVVT-1001
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Translation Pattern
0
2001
Translates “0” to
2001 and Forces a
Second Lookup
Ext. 2001
71
Translation Pattern
Incoming IP WAN Calls Example
Calling Search
Spaces
Partitions
One TP per Unique
DID Range
“E164_Translate”
WAN
“Incoming”
121255.5XXXX [Discard PreDot]
140852.6XXXX [Discard PreDot]
121555.5XXXX [Discard PreDot]
Translation Pattern Must Match the Incoming Called Number
“Internal”
51000
51001
“Internal Only”
64000
Each TP Can Designate a
Different Resulting CSS
© 2009 Cisco Systems, Inc. All rights reserved.
...
TECVVT-1001
64001
Cisco Public
72
Agenda
Reference UC Architecture
Call Processing
Endpoints and Basic Call
Gateways
Media Resources
Call Admission Control
Dialplan
Features
TECVVT-1001
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73
Extension Mobility
Functionality
 Extension Mobility (EM) is an
application that allows a user
to temporarily take ownership
of a phone
 User-specific device profile is
configured for each EM user
and applied to the phone a
user logs in to
 User can log in to any phone
within a Unified CM cluster
that has been enabled for EM
Device Profile
62796
62796
62798
Home
Sue Mobile
TECVVT-1001
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74
Extension Mobility
EM Phone Service Login
Extension Mobility Login/Logout Procedure:
1. User presses Services key on phone
2. Cisco Unified CM returns a list of subscribed services
including Extension Mobility phone service
1
3. User selects and enters UserID and PIN number and
pushes submit to start login process (or selects ‘Yes’
softkey to start logoff)
3
Extension Mobility Login
Extension Mobility
jsmith
2
TECVVT-1001
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75
Mobility and Mobility Applications
Unified Mobility—Mobile Connect, SNR
Call Rings
Remote Destination
408 555-7890
Enterprise
Call to Remote
Destination
408 555-7890
Routed via
Gateway
6
408 555-7890
PSTN
Gateway
Cisco Unified
CM Cluster
5
Remote
Destination:
408 555-7890
PSTN
Call Placed to
Associated Remote
Destination
408 555-7890
1
Dials: 1 408 555-1234
Phone A
2
Remote
Destination
Profile
Call Extended
to Desk Phone
Mobile User’s
Enterprise DN
4
DN: 408 555-1234
Call Extended to
Remote Destination
Profile
DN: 408 555-1234
Shared Line
3
Call to mobile user’s Enterprise directory number rings at desk phone
and Remote Destination phone:
Note: No Changes Are Required on
 Call can be answered at either phone
Mobility User’s Remote Destination Phone
 Once answered all other call legs are cleared
TECVVT-1001
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76
Mobility and Mobility Applications
Unified Mobility—Remote Destination and Desk Phone Pickup
Remote
Destination
Pickup
Call Rings Remote Phone;
Once Answered, Call
Continues Uninterrupted
Between Caller and
Remote Phone
User Presses Mobility
Softkey and then
Select Softkey to
Pickup on Remote
Destination Phone
3 PSTN
Gateway
5551234
MobileConnect On
jsmith
2
408 555-7890
PSTN
DN: 408 555-1234
1
Phone A
Enterprise
Mobile Connect
Call Answered
and in Progress
at Desk Phone
Once Mobile Connect call
is in progress there are
two types of pickup:
1. Remote Destination Pickup:
Mobile user can pickup inprogress desk phone call at
Remote Destination phone
2. Desk Phone Pickup: Mobile
user can pickup in-progress
remote phone call at
TECVVT-1001
© 2009 Cisco Systems, Inc. All rights reserved.
Desk Phone
Pickup
Mobile Connect
Call Answered and
in Progress at
Remote Destination
Upon Remote Destination
Hang Up (or Mid-Call Hold)
User Can Pickup at
Desk Phone by Pressing
the Resume Softkey
1
PSTN
Gateway
408 555-7890
2
PSTN
3
DN: 408 555-1234
Enterprise
Phone A
Cisco Public
Call Continues
Uninterrupted
Between Caller
and Desk Phone
77