1. Unified Communications Design and Deployment
WE ARE EXCITED! To kick off Cisco Live 2009 with you here this Monday morning at 8am. We hope you are too - we most certainly appreciate all of your time and the time your companies give you to come here and attend, so for myself and on behalf of my team here today, we thank you for that. This morning we are going to talk about the CORE components and features of a Cisco UC solution - we have a lot of good information to cover, so we'll buckle up and jump right in.

2. Agenda Reference UC Architecture Call Processing
Endpoints and Basic Call
Gateways
Media Resources
Call Admission Control
Dialplan
Features 2 2

3. Reference UC Architecture
Remote Sites
-30 Users Max per site
Regional Office
-700 Users M M M
MPLS M M
DS3
DS3 M M
HQ in Dallas, TX
Larger Regional office in Chicago
13 remote sites
3000 people in Dallas, 700 people in Chicago, 30 per remote office. Total = ~4000 users.
MPLS Connected. HQ=DS3, Chicago=DS3, Remotes – variable, 1-2 T1 minimum?
If need to show another cluster – go with the hypothetical of buying a large company in Canada? Or South America? (which would afford another cluster?) M M
Headquarters
-3000 Users M
TECVVT-1001
© 2009 Cisco Systems, Inc. All rights reserved.
Cisco Public 3 3 3

4. Network Infrastructure Recommendations for Unified Communications
Unified CM
Cluster
PSTN
SRST
Router
Multiple Qs
802.1p/Q
Link Efficiency (LFI, cRTP)
Classification
Reclassification
Branch Router
Inline Power
Multiple Qs
802.1p/Q
Branch Switch
Inline power
Multiple Qs
802.1p/Q
Fast link Convergence
Campus Access
Multiple Qs
802.1p/Q
Classification
Reclassification
Campus
Distribution
Multiple Qs
802.1p/Q
Traffic Shaping
Link Efficiency (LFI, cRTP)
Classification
Reclassification
WAN
Aggregation
Bandwidth
Provisioning
IP WAN
Campus
Branch Office 4

5. 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? 
Design Zone for Unified Communications

6. Agenda Reference UC Architecture Call Processing
Endpoints and Basic Call
Gateways
Media Resources
Call Admission Control
Dialplan
Features 6 6

7. 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. 7 7

8. 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
Unified CM:
Large Business (30,000 phones per cluster)
Media Convergence Server (MCS) platform
Unified CMBE:
Small – Medium Business (575 phones per server)
No clustering
Co-Res with Unity Connection Voice Messaging
Unified CME:
Small – Medium Business (240 phones per 3845 ISR)
IOS Router platform
Co-Res with Cisco Unity Express (NM or SIM) in chassis 8 8

9. 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
MCS Servers are IBM or HP OEM products.
Publisher holds the master copy of the database, and is the first server created in a cluster. 9 9

10. Call Processing Unified CM Clustering: DB Replication and ICCS
Publisher
Database (DB) Replication DB
Call Processing Servers (Max. 8) DB DB
ccm.exe DB
ccm.exe
ICCS
CTI Manager DB
MoH Server DB DB DB
ccm.exe
ccm.exe
The configuration database is stored on a publisher server, and a read-only copy is replicated to the subscriber members of the cluster. Most of the database changes are made on the publisher and are then communicated to the subscriber databases, thus ensuring that the configuration is consistent across the members of the cluster and facilitating spatial redundancy of the database.
A cluster may contain as many as 20 servers, of which a maximum of eight may run the Unified Communications Manager Service that provides call processing.
Intra-Cluster Communication Signaling (ICCS), involves the propagation and replication of run-time data such as registration of devices, locations bandwidth, and shared media resources. This information is shared across all members of a cluster running the Unified Communications Manager Service, and it ensures the optimum routing of calls between members of the cluster and associated gateways.
IDS = Informix Dynamic Server
TFTP Server DB
Software Conferencing
MS-SQL/IDS Subscribers (Max. 19)
Unified CM 4.x: DB=MS-SQL | OS=MS W2K Server
Unified CM 5.x+: DB=IBM-IDS | OS=Linux 10 10

11. 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) 11 11

12. Call Processing Unified CM Clustering Capacities
Unified CM Approximate Sizing
Server Platform
Maximum Users per Server
High Availability Server
MCS 7845
7500
Yes
MCS 7835
2500
MCS 7825
1000 No
Cisco Unified Communications Sizing Tool
Accessible to Cisco and Partners only
Good for you to know about! 12 12

13. 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
Each model builds on each other.
Call deployment, but call out as CALL PROCESSING models. Other types of deployment models (VM, IPCC, etc.), but this is CP. 13

14. Unified CM Deployment Models Single Site
Unified CM, applications and DSP resources at same physical location
Supports up to 30,000 SIP or SCCP phones per cluster
PSTN used for all external calls
Applications (VMail, IPCC, MP…)
Unified CM Cluster
Single Site is the base model.
PSTN 14 14 28

15. Deployment Models Centralized Call Processing
Applications (VMail, IPCC, MP…)
SRST-Enabled
Routers
PSTN
Unified CM
Cluster
Branch A
IP WAN
Headquarters
Centralized takes the Single Site model, and adds IP Phones over a WAN, which opens up more design aspects to consider.
Unified CM cluster at central/HQ site
Applications and DSP resources can be centralized or distributed
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)
Branch B 15 15 28

16. Deployment Models Distributed Call Processing (Unified CM-Unified CM Model)
Applications (VMail, IPCC, MP…)
Applications
PSTN
Unified CM
Cluster
Unified CM
Cluster GK
IP WAN
Regional Branch A GK
Headquarters
Gatekeeper
Applications
Distributed Call Processing Model is the incorporation of multiple Singe Site Models. The INTERcluster communication opens up yet more design aspects to consider.
Remember 30K phones per cluster, so in this example, could theoretically have 90K phones.
Unified CM, applications, and DSPs located at each site
Up to 60,000 SIP or SCCP phones per cluster
~100 sites
Transparent use of PSTN if IP WAN unavailable
Each cluster can be single site or centralized call processing topology
Unified CM
Cluster
Regional Branch B 16 16 28

17. Deployment Models Clustering over the WAN (CoW)
Unified CM Cluster
Voice Mail
Voice Mail
Distance
San Jose
San Francisco
CoW involves a Single Site model, where Subscribers are displaced over a WAN. This also opens up serious design aspects that need consideration.
Bandwidth requirement for CoW with 6.1 has increased as well
See the Unified Communications Solution Reference Network Design (SRND) based on Unified CM 6.1, Unified Communications Deployment Models Chapter:
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
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
Increased to
80-ms RTT
in 6.1
B/W Required Increased in 6.1 17 17

18. Unified CM Call Processing Failover and Redundancy
Directory Services
Gateways
Music on Hold
Software Conferencing
Unified CM Subscriber
Software MTP
DSP Resources
Conferencing
TFTP
Conf
Failed
Call Processing
CTI/QBE I/F
SCCP I/F
MGCP I/F
DSP Resources
Transcoding
Xcode
H.323 I/F
Intra-Cluster Communications (ICCS)
SIP I/F
Directory Services
Cisco Unity Vmail Server
Controlled by the Device Pool’s CM GROUP configuration, or ability of application to designate a primary/secondary Unified CM.
Music on Hold
Software Conferencing
Unified CM Subscriber
Software MTP
JTAPI
IP-IVR
TFTP
Call Processing
CTI/QBE I/F
SCCP I/F
IP Phones
Active Unified CM Server
MGCP I/F
H.323 I/F
SIP I/F 18 18

19. Unified CM Redundancy 1:1 vs. 2:1 Redundancy
MCS 7835 Supports 2500 Phones/Server
2:1 Redundancy Scheme
1:1 Redundancy Scheme
1 to 1250
1251 to
2500
1 to
2500
Backup
2501 to
5000
2501 to
3750
3751 to
5000
Cost-efficient redundancy
High Availability during upgrades
Maximum of 10,000 backup registrations/server
Load-sharing redundancy
High Availability during upgrades
Faster failover
Remember only 8 Call Processing Nodes allowed. 19 19

20. Unified Communications Infrastructure Failover and Redundancy: 1:1 Redundancy Example
Primary
Backup
MCS 7845 supports 7500 phones/server
Load-share between primary and backup servers
Backup
Primary
Phone Set 1
Phone Set 2
To 7,500 IP Phones
To 15,000 IP Phones
To 30,000 IP Phones
Publisher and
TFTP Server(s)
Publisher and
TFTP Server(s)
Publisher and
TFTP Server(s) 1–
3750
3751 to
7500 1–
3750
3751–
7500
1 to 3750: Primary
3751 to 7500: Backup
7501–
11,250
11,251–
15,000
3751 to 7500: Primary
1 to 3750: Backup
7501–
11,250
11,251–
15,000
15,001–
18,250
18,251–
22,500
22,501–
26,250
26,251–
30,000 20 20

21. Unified Communications Infrastructure Failover and Redundancy: Survivable Remote Site Telephony
WAN Failure
Normal Operation
Unified CM
Cluster
Signaling Traffic
Signaling Traffic
Voice Traffic
IP WAN
SRST
Capable
Router
Voice Traffic
Applications
Branch Site
PSTN
Central Site
With SRST, important to remember that it is a subset of the features available (vs. Unified CM), but these have grown far beyond just simple dial tone and PSTN accessibility. CCO can provide a list of the latest features. There is even CME as SRST which offers even more functionality (like ACD, nightbell, etc.)
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 dial-peers; MGCP GWs require the ‘MGCP Fallback to H323’ feature 21 21

22. Unified SRST Configuration Example
voice service voip
allow-connections sip to sip
sip
bind control source-interface GigabitEthernet0/0.300
bind media source-interface GigabitEthernet0/0.300
registrar server expires max 600 min 60 !
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729br8
voice register global
max-dn 20
max-pool 20
external-ring bellcore-dr4
voice register pool 1
id network mask
translate-outgoing called 1
voice-class codec 1
call-forward b2bua mailbox
call-forward b2bua busy
call-forward b2bua noan timeout 10
voice translation-rule 1
rule 1 /^4\(...\)/ / \1/
voice translation-profile profile1
translate called 1
dial-peer voice 90 pots
destination-pattern
port 1/0/0
dial-peer voice 100 voip
destination-pattern ….
monitor probe icmp-ping
session protocol sipv2
session target ipv4:
session transport tcp
incoming called-number .T
dtmf-relay rtp-nte !
call-manager-fallback
access-code fxo 9
default-destination pattern 2001
ip source-address port 2000
keepalive 30
max-ephones 24
max-dn 48
   voic
translation-profile outgoing rule1
   call-forward busy
   call-forward noan timeout 10
SCCP SRST Configuration
SRST 3.4 or later supports SIP and SCCP IP phones simultaneously and it requires 12.4(6)T or later
SRST 3.4 or later operates as a B2BUA mode for SIP phones
SIP SRST Configuration 22 22

23. Agenda Reference UC Architecture Call Processing
Endpoints and Basic Call
Gateways
Media Resources
Call Admission Control
Dialplan
Features 23 23

24. 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

25. PORTFOLIO AVG LIST PRICE ($USD)
思科IP电话终端一览
9900 Series
9900 Series
Higher
Voice/
Video/
Apps
8900 Series
8900 Series
SPA 500
Series
PORTFOLIO AVG LIST PRICE ($USD)
7900 Series
7900 Series
7900 Series
7900 Series
IP PHONE APPLICATIONS
6900 Series
6900 Series
6900 Series
This provides a view of the entire Cisco Unified IP Phone portfolio. The “Y” axes scales are for context and should not be taken as an absolute scale for any of the columns.
SPA 500
Series
SPA 900
Series
3900 Series
3900 Series
Voice
Lower
3900 Series
Unified
CME on ISR
SPA 9000
UC-500
Unified CMBE
Unified CM 25 25

26. Unified Communications Core: Network Services
IP Phone Boot-Up Process
Inline Power (ILP)
Inline Power Initialization
Cisco Discovery Protocol (CDP) or Link Layer Discovery Protocol-Media Endpoint Discovery (LLDP-MED)
ILP Negotiation, Voice VLAN ID
Dynamic Host Configuration Protocol (DHCP)
IP Assignment, TFTP Server Allocation, DNS (optional)
Trivial File Transfer Protocol (TFTP)
Configuration File, IP Phone Firmware
To illustrate the Network Services required for your UC Infrastructure, let’s focus on the process an IP Phone follows when plugged into the Network Infrastructure. 26 26

27. Unified Communications Core: Network Services
DHCP
Inline Power Provided
Cisco Catalyst Switch
CDP/LLDP Neighbored
DHCP Req
DHCP Server
DHCP Rsp (IP Add, Def-GW, TFTP, DNS*)
Option 150 or Option 66
DHCP Request Must Be Made in
the Correct VLAN to Place the
Phone in the Correct Subnet!!
Sends a DHCP Request to the DHCP Server via a broadcast on the Voice VLAN it received via CDP
If a Static IP Address is configured on the Phone, Gateway address then no DHCP Server request is made.
DHCP Server sends the IP address, mask, DNS, Default Gateway, and TFTP Server Address
THERE ARE FOUR WAYS THE TFTP server address can be obtained:
Static address configured on the Phone (Option 8, Option 32 Alternate TFTP must be set to Yes)
Option 150 (single IP address) from the DHCP Server
Option 66 (first IP address or DNS name)
Look up CiscoCM1.your.domain __ ONLY if no TFTP option is given via DHCP.
Phone displays: “Configuring IP” (DNS is optional)
Phone settings: Settings=>NetCfg=>“DHCP Server” Settings=>NetCfg=>“IP Address” Settings=>NetCfg=>“TFTP Server X” 27 27

28. Unified Communications Core: Network Services
TFTP
Unified CM
Cluster
UCM1
MAC Address:
001956A6A7ED
UCMx
Registration (SCCP, SIP)
UCM2
Backup Link
Publisher
TFTP
TFTP: GET Configuration File(s) for MAC
Get configuration from Unified CM TFTP. This configuration file is in the XML format. Once the XML file is retrieved, it parses the file and reads load ID and stores the version stamp values. It compares the loadID with its own and if there is a mismatch, the phone retrieves the new load file from TFTP Server and then resets. If there is a match, the phone goes ahead with registration.
Phone Configuration, Firmware Download (If Required)
1=UCM1:
2=UCM2: …
CM Group: UCM1
UCM2
Device Pool 28 28

29. 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
IP Phone A
Signaling Leg 1
Unified CM
Media
IP WAN
However, with this deployment model, as we discussed before, there are now new design considerations as far as where to locate DSP resources, how to handle IP WAN congestion & outages, branch scalability.
Dial Plan Lookup
Signaling Leg 2
IP Phone B 29 29

30. Unified Communications Core: Network Services
Signaling Protocols
Carriers/
Other Vendors PBXs
Unified CM 5.x+
Unified CME
Gateways
Conf/ Xcode
Unified CM 5.x+
DSP Resources
Cisco Unified Presence Server
Rich-Media Conferencing
Soft Phones
Unified Messaging
Microsoft LCS/OCS
SIMPLE (Session Initiation Protocol for Instant Messaging and Presence Leveraging Extensions)
IBM Sametime
SCCP
MGCP
H.323
CTI
SIP/SIMPLE
CSTA over SIP
Cisco Unified Personal Communicator
CTI Apps
Cisco and 3rd-party Phones
Video Endpoints
Cisco Unified Communications provides seamless interworking with SIP, H.323, MGCP, SCCP, TAPI/JTAPI and Q.SIG Protocols 30 30

31. Unified Communications Core: Network Services
Signaling Protocols: Unified CM as “Protocol Translator”
Session Initiation Protocol
IP Phones
Telepresence
Skinny Client Control Protocol
ITU-T H.323 Standard
SIP Networks
IP Phones
Gateways
Analog
Phones
Wireless IP Phones
SIP
Gateways
Analog Phones
PC-Based IP Phones
Video
Terminals
SCCP
H.323
You can think of Unified CM as a “Protocol Translator” allowing communications across all.
Computer Telephony Integration/ Quick Buffer Encoding
Media Gateway Control Protocol
Gateways
Analog
Phones
MGCP
CTI/QBE
Applications
Servers
(JTAPI/CTI)
Call Agents 31

32. 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
Note that a dedicated TFTP server (and a backup) is recommended for over 1250 users

33. 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
TECVVT-1001
© 2009 Cisco Systems, Inc. All rights reserved.
Cisco Public 33 33 33

34. Agenda Reference UC Architecture Call Processing
Endpoints and Basic Call
Gateways
Media Resources
Call Admission Control
Dialplan
Features 34 34

35. 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

36. Unified Communications Core: Gateways Gateway Selection Criteria
Cisco Unified CM
PSTN
IP WAN
Router/
Gateway
Voice port density requirements
Signaling protocol (H.323, MGCP, SIP, etc.)
Support for required PSTN signaling types
Support for required WAN interfaces and QoS 36 36

37. Unified Communications Core: Gateways H.323
TDM IP
PRI Layer 3
PSTN
Layer 2
H.225
Framing
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 37 37

38. These Capabilities Do Not Exist for MGCP-Controlled GWs
Unified Communications Core: Gateways The Power of Cisco IOS Dial-Peers: H.323 and SIP
dp 10 pots
dp 1 voip IP
PSTN
dp 11 pots
dp 2 voip
dp 12 pots
dp 3 voip
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
These Capabilities Do Not Exist for MGCP-Controlled GWs 38 38

39. Unified Communications Core: Gateways Protocol and Platform Recommendations
BRKVVT-2010
Designing UC Gateways and DSP Engineering in Enterprise Networks
Cisco
Unified CM
Cluster
PSTN
H.323, SIP, MGCP fallback to H.323/SIP
Standalone, Router-integrated
Platforms:
17XX, 18XX
26xx, 28XX
37xx, 38xx
SRST
Router
MGCP, SIP, H.323
Standalone, Router-integrated
Platforms:
WS-X6608, CMM
26XX, 28XX
37XX, 38XX
Router/
Gateway
IP WAN
Central Site
Remote Site 39 39

40. 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
Mine
Yours
Security
H.323 and SIP
SIP Normalization
DTMF Interworking
Transcoding
Codec Filtering
Fax/Modem Support
Encryption
Authentication
Registration
SIP Protection
FW Placement
Toll fraud

41. Unified Communications Core: Cisco Unified Border Element
Usage in IP-PSTN Connectivity
Cisco Unified Communications Manager Cluster
Cisco Unified Border Element
IP-PSTN Service Provider
H.323/SIP Trunk
SBC
SIP
Trunk
Network/Topology Hiding for Voice and Video Calls
Protocol Support—H.323 and SIP
Voice Codecs—G.711, G.729, G.726, G.723, G.728, Transparent
Video Codecs—H.261, H.263 and H.264
Codec Filtering
Media—Media Flow Through and Media Flow Around
DTMF Interworking—H.245 Alphanumeric, Signal, RFC2833, SIP NOTIFY
Fax/Modem—T.38, Passthrough, Cisco Fax Relay, Modem Passthrough
Security—TLS, IPSec with SRTP
Signaling Interworking
Supplementary Services
Transcoding
Transport Mode—TCP, UDP
Number Translation
Quality of Service
Call Admission Control
Call Detail Records
TCL/VXML Support
Rotary Support 41

42. Unified Communications Core: Cisco Unified Border Element
BRKVVT-2305
Integrating Voice and Video over IP Networks Using the Cisco Unified Border Element
Address Hiding
CUBE
CUBE
Site A
Unified Border
Element
IP WAN
Site B
Unified Border
Element
IP WAN
Cisco Unified CM Cluster
Cisco Unified CM Cluster
Site A— x/24
x/24
Site B— x/24
Within the same company—between departments having overlapping addresses
Integrating new acquisition into the existing voice network 42 42

43. Agenda Reference UC Architecture Call Processing
Endpoints and Basic Call
Gateways
Media Resources
Call Admission Control
Dialplan
Features 43 43

44. 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

45. Unified Communications Core: Media Resources
Conferencing, Transcoding, Music on Hold
Conference Bridge
DSPs needed for multi- codec conferences
Media Termination Point
Media Termination
DSPs optional
Transcoding
DSPs needed to transcode multiple CODEC types (e.g., G.711 to G.729)
Automatic codec selection
Music on Hold
Multiple source types possible (centralized or branch-based)
IVR
Cisco
Unified CM
Cluster
Music on Hold
Transcoder
Xcode
MTP
MTP
Conference
Bridge
Conf
PSTN
Conference bridges mix audio between multiple endpoints involved in a conference.
Media Termination Points extend supplementary services, such as call hold, call transfer, call park, and conferencing, that are otherwise not available when a call is routed to an H.323 endpoint. Some H.323 gateways may require that calls use an MTP to enable supplementary call services, but normally, Cisco IOS gateways do not.
A transcoder takes the stream of one codec and transcodes (converts) it from one compression type to another compression type. For example, it could take a stream from a G.711 codec and transcode (convert) it in real time to a G.729 stream. In addition, a transcoder provides MTP capabilities and may be used to enable supplementary services for H.323 endpoints when required.
MoH provides a stream of audio to a device that is being held (due to a transfer or being placed on hold)
IP WAN
... 45 45

46. 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
MRM = Media Resource Manager
MRG = Media Resource Group
MRGL = Media Resource Group List
1st
Choice
2nd
Choice
Media Resource Group
Media Resource Group
2nd
Choice
1st
Choice
2nd
Choice
1st
Choice
Media Resource 1
Media Resource 2
Media Resource 3
Media Resource 4 46 46

47. Unified Communications Core: Media Resources
MRGL and Device Association
Site A
MRGL—A
Device Pool—A
CM Group
Date/Time Group
Region
Media Resource Group List
MRG—A
Media
Resources
Assign a MRGL Directly to the Device Take a Higher Priority than Device-Pool Based Configuration
Site B
MRGL—B
Device Pool—B
CM Group
Date/Time Group
Region
Media Resource Group List
MRM = Media Resource Manager
MRG = Media Resource Group
MRGL = Media Resource Group List
MRG—B
Media
Resources
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 47 47

48. Unified Communications Core: Media Resources
Centralized vs. Distributed DSPs
Cisco Unified CM Cluster X
PSTN A
Centralized
DSPs
IP WAN
Central
Site B
Conf
Branch
$ Bandwidth vs. $ Hardware
Cisco Unified CM Cluster X
PSTN A
Distributed
DSPs
IP WAN B
MRG
Conf
MRG
Conf
MRG
Conf
Branch
Central
Site 48

49. 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)
CFB=Conference Bridge
MTP=Media Termination Point
XCODE=Transcoder
MOH=Music on Hold
MRG=Media Resource Group
MRGL Media Resource Group List
MRM=Media Resource Manager 49

50. Unified Communications Core Media Resources
dsp
dsp
dsp
dsp
dsp
MPLS
dsp
dsp
dsp
dsp
dsp
dsp
HQ in Dallas, TX
Larger Regional office in Chicago
13 remote sites
3000 people in Dallas, 700 people in Chicago, 30 per remote office. Total = ~4000 users.
MPLS Connected. HQ=DS3, Chicago=DS3, Remotes – variable, 1-2 T1 minimum?
If need to show another cluster – go with the hypothetical of buying a large company in Canada? Or South America? (which would afford another cluster?)
dsp
dsp
dsp
TECVVT-1001
© 2009 Cisco Systems, Inc. All rights reserved.
Cisco Public 50 50 50

51. Agenda Reference UC Architecture Call Processing
Endpoints and Basic Call
Gateways
Media Resources
Call Admission Control
Dialplan
Features 51 51

52. Call Admission Control Why Is It Needed?
Circuit-Switched
Networks
Packet-Switched
Networks
IP WAN
Router/
Gateway UC
Manager
IP WAN Link’s LLQ Is Provisioned for Two Calls (Equivalent to Two “Virtual” Trunks)
PSTN
Physical
Trunks
IP WAN
Link
No Physical Limitation on IP Links
Third Call Can Go Through, but Voice Quality of All Calls Degrades
 Call Admission Control Blocks Third Call
Third Call
Rejected
This is applicable to video as well. The migration of ISDN base video to IP presents the same issues. High bandwidth video may also present a CAC issue in the campus
STOP
PBX 52

53. Unified Communications Core Call Admission Control
BRKRST-2505
Call Admission Control Design for Unified Communications
BRKVVT-3303
Advanced Call Admission Control Design, Implementation, and Troubleshooting Using RSVP
Two Types of CAC:
Topology-Unaware
Topology-Aware
Three Options for Configuring CAC:
Unified CM Locations (Topology-Unaware)
RSVP (Topology-Aware)
Gatekeeper Zones (Topology-Unaware) 53 53

54. Unified CM Static Locations Concept
Central
Site
Prevent WAN link over-subscription by limiting voice bandwidth
Assign bandwidth limit for voice per location (G729 = 24Kbps, G711=80Kbps)
Location1 makes a G729 call over WAN to the Central Site 1
<Hub_None>
Location
PSTN
IP WAN
Remote
Sites
Location 1
Location 2
Max BW = 24 kbps
Avail BW = 24 kbps
Max BW = 48 kbps
Avail BW = 48 kbps 54

55. Unified CM Static Locations Concept
Central
Site
Prevent WAN link over-subscription by limiting voice bandwidth
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 2nd G729 call, but Locations-based CAC blocks the call
When resources are insufficient, by default the user hears a fast-busy tone and a configurable message is displayed 1
<Hub_None>
Location
PSTN
IP WAN
Remote
Sites 2
STOP
Location 1
Location 2
Max BW = 24 kbps
Avail BW = 0 kbps
Max BW = 48 kbps
Avail BW = 48 kbps 55

56. Unified CM Static Locations Concept
Central
Site
Prevent WAN link over-subscription by limiting voice bandwidth
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 2nd G729 call, but Locations-based CAC blocks the call
When resources are insufficient, by default the user hears a fast-busy tone and a configurable message is displayed 3 1
<Hub_None>
Location
PSTN
IP WAN
Remote
Sites 2
STOP
Automated Alternate Routing (AAR) sends call via PSTN seamless to the user.
Location 1
Location 2
Max BW = 24 kbps
Avail BW = 0 kbps
Max BW = 48 kbps
Avail BW = 48 kbps 56

57. Unified CM Static Locations Bandwidth Provisioning
Provision LLQ PQ
with These Values
CUCM Location
Actual L3 Bandwidth
L2 Bandwidth (Frame Relay)
G.711 Audio
80 Kbps
80 Kbps (64K + Header)
81.6 Kbps (80K + L2 Header)
G.729 Audio
24 Kbps
24 Kbps (8K + Header)
25.6 Kbps (24K + L2 Header)
384K Video
384 Kbps
420 Kbps (384K + Est. L2/L3 Headers
For purposes of Unified CM Location bandwidth calculations only, assume that each call stream using these condecs consumes the following amount of bandwidth:
G > 80 kb/s.
G > 80 kb/s.
G > 24 kb/s.
G > 16 kb/s.
G > 24 kb/s.
GSM --> 29 kb/s.
Wideband --> 272 kb/s. 57 57

58. 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.711-only 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 58 58

59. Call Admission Control
Simple Hub and Spoke Topologies: Centralized Deployments
Use Static Locations: one location per spoke site
Devices at hub site in <Hub_None> location
Up to 1,000 locations per Cisco Unified CM cluster
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
Hub Site
Increased to
2000 in 7.1
Loc.
<Hub_None>
...
Loc. 1
Loc.1000
Spoke Sites 59 59

60. Agenda Reference UC Architecture Call Processing
Endpoints and Basic Call
Gateways
Media Resources
Call Admission Control
Dialplan
Features 60 60

61. 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
Pattern
Configuration Order
Route List
Points to prioritized route groups
Performs digit manipulation (opt)
Route
List
1st Choice
2nd Choice
Route Group
Points to the actual devices
Distribution algorithm
Route
Group 1
Route
Group 2
1st
Choice
2nd
Choice
Digit Manipulation on Route Pattern affects everything below, etc.
Devices
Gateways (MGCP, SCCP, H.323)
Gatekeeper (H.323)
Trunk (H.323, ICT, SIP) GK
IP WAN
PSTN 61 61 28

62. 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! #
is a macro that enters the Whole North American Numbering Plan (NANP) into Cisco Unified CM (166 specific individual Route Patterns).
A Macro That Enters the Whole North American
Numbering Plan into Cisco Unified CM (or a Different Country’s Numbering Plan If Using the International Dial Plan Tool) 62

63. 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!#
Matches Any Number That Begins with 13, Is
Followed by One or More Digits, and Ends with #;
135# and 13579# Are Example Matches 63

64. Cisco Unified CM Call Routing Logic Basic Principle
User Dials
“1200”
Route Patterns
1XXX
12XX
User Dials
“1234”
1234
Directory Numbers
Note, DN 1234 is not chosen due to the fact that it is a Directory Number, but rather because it is the most specific longest-match.
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 64

65. Building Classes of Service Concepts
Phones
“Dialing” Devices
“Dialable” Patterns
PartitionA
CSS1
PartitionA
PartitionB
2002
Lines (Directory Numbers)
2001
2000
Translation Patterns
7 [Transform Mask: 2001]
Lines
911
CSS2
PartitionB
Route Patterns
9.[2-9]XXXXXX
9.[2-9]XX[2-9]XXXXXX
Gateways
PartitionB
Using the LINE in this slide as an example, it is assigned to CSS2 which can ONLY search through PartitionB to find possible matches for dialed numbers. In this example, if that Line goes offhook and dials a “2”, the user would immediately hear an Annunciator Message (“Your call can not be completed as dialed”) since there are no patterns that start with a “2” in PartitionB.
CSS3
PartitionB
PartitionA
Application Numbers
(CTI Route Points, CTI Ports)
5000
900X
Special numbers
(MeetMe, CallPickup...)
99XX
Applications
8001
Voice Mail Ports
CSS4
PartitionA
8000
Route Patterns
9.011! 65

66. Dial Plan Example Single Site Deployment Model: Composite View
Calling Search
Spaces
Route
Lists
Route
Groups
Partitions
Devices
Calling
Search
Space
Assigned
Internal
Route
Patterns
All IP Phone DNs
Internal Only
911
9.911
LOC RL
LOC RG
PSTN
Local
Local
9.[2–9]XXXXXX 2
9.[2–9]XX[2–9]XXXXXX
Remember, Partition and CSS names are user-defined.
This example illustrates having a 2nd PRI with better long distance rates.
National
National
9.1[2–9]XX[2–9]XXXXXX LD RL LD RG 1
International
International
9.011!
9.011!# 66 66

67. Dial Plan Example Centralized Deployment Model: Composite View
Calling Search
Spaces
Route
Lists
Route
Groups
Partitions
Devices
Calling
Search
Space
Assigned
NYC911
911
9.911
Route
Patterns
NYCInternal
NYCPSTN
9.[2–9]XXXXXX
9.1[2–9]XX[2-9]XXXXXX
9.011!
9.011!#
NYC
PSTN
NYC
PSTN
PSTN
NYC
Phones
NYCAllCalls
NYC
Gateways
Internal
All IP Phones
PHLInternal
PHL911
911
9.911
PHL
PSTN
PHL
PSTN
PSTN
PHLPSTN
9.[2–9]XXXXXX
9.1[2–9]XX[2–9]XXXXXX
9.011!
9.011!#
PHLAllCalls
PHL
Gateways
PHL
Phones 67 67

68. Dial Plan Example Centralized Deployment Model: Composite View
~ Using Local Route Group feature of Unified CM version 7.0 ~
Calling Search
Spaces
Route
Lists
Route
Groups
Partitions
Devices
Calling
Search
Space
Assigned
NYCInternal
Internal
All IP Phones
PSTN
NYC
Phones
NYCAllCalls
US EMERG
911
9.911
NYC
Gateways US
PSTN
Local Route Group
US PSTN
9.[2–9]XXXXXX
9.1[2–9]XX[2-9]XXXXXX
9.011!
9.011!#
PHLInternal
PSTN
PHL
Gateways
PHLAllCalls
Route group chosen as per device pool of calling device
PHL
Phones
Route
Patterns 68 68

69. 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[1-6]XXXXXXXX, 000!, 000!#
We get site-specific failover for “free” on long distance patterns
We now have much fewer things to configure per site

70. 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 70 70 70

71. 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
Translation Pattern
Translates “0” to
2001 and Forces a
Second Lookup
Ext. 2001 71 71

72. Translation Pattern Incoming IP WAN Calls Example
Calling Search
Spaces
Partitions
One TP per Unique
DID Range
“E164_Translate”
“Incoming”
XXXX [Discard PreDot]
WAN
XXXX [Discard PreDot]
XXXX [Discard PreDot]
Translation Pattern Must Match the Incoming Called Number
Note that the translation patterns have to be tailored to whatever the LEC delivers as a called party number into each gateway.
“Internal”
51000
“Internal Only”
51001
64000
Each TP Can Designate a Different Resulting CSS
64001
... 72 72

73. Agenda Reference UC Architecture Call Processing
Endpoints and Basic Call
Gateways
Media Resources
Call Admission Control
Dialplan
Features 73 73

74. 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
The Extension Mobility feature dynamically configures a phone according to the authenticated user's device profile. The benefit of this application is that it allows users to be reached at their own extension on any phone within the Unified CM cluster, regardless of physical location.
Uses for EM:
Locking phones down
Providing user mobility
Deployment of Phone
62796
62798
Home
Sue Mobile

75. Extension Mobility EM Phone Service Login
Extension Mobility Login/Logout Procedure:
User presses Services key on phone
Cisco Unified CM returns a list of subscribed services including Extension Mobility phone service
User selects and enters UserID and PIN number and pushes submit to start login process (or selects ‘Yes’ softkey to start logoff) 1
Extension Mobility
Extension Mobility Login
jsmith
How many simultaneous extension mobility logins can the CUCM handle? In 5.1(2), a 7845 server can do 90 logins a minute. Prior to that, in 5.0, it could only to 50 logins per minute 3 2

76. Mobility and Mobility Applications Unified Mobility—Mobile Connect, SNR
Call Rings
Remote Destination 6
Enterprise 5
Call to Remote
Destination
Routed via
Gateway
Cisco Unified
CM Cluster
PSTN
Gateway
Remote
Destination:
PSTN
Call Placed to
Associated Remote
Destination 4 1
Dials:
Call Extended
to Desk Phone 2
Phone A
Remote
Destination
Profile 3
Call Extended to
Remote Destination
Profile
Mobile Connect, also known as Single Number Reach, provides Cisco Unified Communications users with the ability to be reached via a single enterprise phone number that rings on both their IP desk phone and their cellular phone simultaneously
Mobile Connect users can pick up an incoming call on either their desk or cellular phones and at any point can move the in-progress call from one of these phones to the other without interruption
Mobile User’s
Enterprise DN
DN:
DN:
Shared Line
Call to mobile user’s Enterprise directory number rings at desk phone and Remote Destination phone:
Call can be answered at either phone
Once answered all other call legs are cleared
Note: No Changes Are Required on Mobility User’s Remote Destination Phone 76

77. Mobility and Mobility Applications Unified Mobility—Remote Destination and Desk Phone Pickup3
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 2
MobileConnect On
jsmith
PSTN
Gateway
PSTN
DN: 1
Mobile Connect
Call Answered
and in Progress
at Desk Phone
Enterprise
Desk Phone
Pickup 1
Mobile Connect
Call Answered and
in Progress at
Remote Destination
Phone A
Upon Remote Destination
Hang Up (or Mid-Call Hold)
User Can Pickup at
Desk Phone by Pressing
the Resume Softkey 2
Once Mobile Connect call is in progress there are two types of pickup:
PSTN
Gateway
Remote Destination Pickup: Mobile user can pickup in-progress desk phone call at Remote Destination phone
Desk Phone Pickup: Mobile user can pickup in-progress remote phone call at desk phone
PSTN
DN: 3
Call Continues
Uninterrupted
Between Caller
and Desk Phone
Enterprise
Phone A 77