Designing An Enterprise IPT Network

Designing an Enterprise
IP Telephony Network
Andres Martinez
© 2003, Cisco Systems, Inc. All rights reserved. 1

Agenda
• Introduction
• Network Infrastructure
• Telephony Infrastructure
• Applications

Focus of This Session
PBX PBX
IP WAN
Router/GW Router/GW
Toll Bypass
CallManager Applications
Applications CallManager
IP WAN
Router/GW Router/GW
End-to-End IP Telephony with Applications

Scope of This Seminar
CallManager Applications
Applications CallManager
IP WAN
Router/GW Router/GW
• Understanding what can be built today

• Learning how to build it
• To find out more about IP telephony design:
http://www.cisco.com/go/srnd/

The Big Picture:
End-to-End IP Telephony
Telecommuter Applications
Call
Manager Rest of the World
Cluster
Gatekeeper
GK
Headquarters
Road V3PN
Warrior
IP WAN PSTN
Internet
Large
Branch Legacy
Office Site
Tie
Survivable Remote IOS Telephony Lines
Site Telephony Service
Small PBX
Branch
Offices
The Elements of IP Telephony
Voice Mail/ Auto-Attendant/ XML LDAP
Unified Interactive
Voice Response
Phone
Services
Directory Applications
Messaging
Telephony Media Call Processing PSTN Telephony Infrastructure

Endpoints Resources Agents Gateway
Gateway/ Telephony
GK Survivable Endpoints
Remote
Xcode
PSTN
Conf
Si
IP WAN
Si Branch Access
Router Switch Branch
Access Distribution/ WAN
Switch Core Switch Aggregation
Campus
Router Network Infrastructure
Agenda
• Introduction
• Applications

Network Infrastructure Agenda
• Building a Campus Network

• Enabling QoS in the Campus
• Overlaying Wireless LANs
• Building a WAN
• Enabling QoS in the WAN

Building a Campus Network
Multi-Layer Network Design
Access
Layer 2
Distribution
Layer 3
Core
Layer 3
Distribution Building
Layer 3 Block
Access = L2 Links
Layer 2 = L3 Links
Server Farm
VLAN Model
2 3 4 5 6 7 8 9 Unique VLAN 20 21 22 23 etc.
Access
Layer
.......
in Each
Wiring Closet
Wiring Wiring Switch Wiring Wiring
Closet Closet Closet Closet
• A VLAN = an IP subnet
• VLANs do not span different wiring closet switches
with a few exceptions
• If 2+ VLANs per access switch, load sharing is very
easy to achieve
• This model achieves fast convergence and high
stability
Overlaying Wireless LANs
VLAN Design
• Create a single VLAN for

the wireless LAN per VVID=110 VLAN=10
campus building
• Need a L2 link between VVID=111 VLAN=11
distribution switches to
Si
carry the wireless VLAN
WLAN=30
• Spanning tree
convergence only
Si
affects the WLAN VVID=120 VLAN=20
• Layer 2 roaming within

the building (Layer 2 VVID=121 VLAN=21
domain spans multiple
wiring closet switches) Distribution Access
Layer Layer
WLAN=30
Distribution Layer
Distribution HSRP HSRP

Layer Features: Primary Primary
VLAN 110 VLAN 10
• Passive
interface
default
Cat6000 Cat6000 Cat35xx Cat35xx Cat35xx
• HSRP, HSRP VVID=110 VVID=111 VVID=310 VVID=311 VVID=312

Track/Preempt
• OSPF/EIGRP:
Adjust timers
Summary
address VLAN=10 VLAN=11 VLAN=30 VLAN=31 VLAN=32
Path costs Model #1 Model #2

Core Layer
Core Layer Features:

Core
• Each link belongs to
its own /30 subnet
• No STP in the core—
Distribution
All routed
• Load balancing to
the core/server farm Cat6000 Cat6000 Cat35xx Cat35xx Cat35xx
by default VVID=110 VVID=111 VVID=310 VVID=311 VVID=312
• Tune routing
protocol timers for
fast convergence VLAN=10 VLAN=11 VLAN=30 VLAN=31 VLAN=32
Model #1 Model #2
Summary
• Access Layer
Access
Per-VLAN Layer 2
spanning-tree
Rootguard
portfast Distribution
UplinkFast
Layer 3
• Distribution Layer Server Farm
HSRP with Core
load balancing Layer 3
OSPF/EIGRP
configured for fast Distribution
convergence Layer 3
• Core
OSPF/EIGRP Access
configured for fast Layer 2
convergence
WAN Internet PSTN

Enabling QoS in the Campus
Congestion Scenario: TCP Traffic Burst + VoIP
Instantaneous
Core Interface
Si Si
Typical 4:1 Congestion
Data Over-
Subscription
Distribution
Si Si
Typical 20:1
Data Over-
Subscription Access
= Data
= Voice

Enabling QoS in the Campus
Cisco’s Approach to QoS
Classification: Mark the Packets with a Specific Priority Denoting a
Requirement for Class of Service from the Network
Trust Boundary: Define and Enforce a Trust Boundary at the Network Edge
Scheduling: Assign Packets to One of Multiple Queues (Based on

Classification) for Expedited Treatment through the Network
Provisioning: Accurately Calculate the Required Bandwidth

for All Applications Plus Element Overhead
CallManager
Cluster
PSTN
SRST
Router
IP WAN
Campus Branch Office

WLAN QoS Challenges
Downstream Downstream
Network
Upstream Upstream
Bi-Directional QoS Bi-Directional QoS
• Downstream traffic from AP receives “soft QoS” as

VoIP receives preferred treatment, but not a strict
priority queue
• Upstream traffic from Cisco 7920 uses enhanced
access to RF medium, so it will also get “soft QoS”

WLAN Sizing—
Number of Concurrent CALLS per AP
• G.711—7 concurrent calls

• G.729—8 concurrent calls
• Call numbers are with Voice Activity
Detection (VAD) disabled
(VAD is a global parameter on CM)
• 20ms sampling rates on phones 100pps
(full-duplex) of Real Time Protocol (RTP)

QoS and WAN Considerations
WAN Topologies and Technologies
Central Central
Site Site
Leased Lines
Frame Relay
ATM
FR/ATM SIW
IPSec V3PN
MPLS
QoS- QoS-
Enabled L2 WAN L3 WAN Enabled
Branch 1 Branch 2 Branch 3 Branch 1 Branch 2 Branch 3
Hub and Spoke Full Mesh

Enabling QoS in the WAN
Elements that Affect End-to-End Delay
CallManager
Cluster
PSTN
SRST
Router
IP WAN
Campus Branch Office
Propagation
CODEC Queuing Serialization Jitter Buffer
and Network
Variable Variable 6.3 µs/Km +

G.729A: 25 ms (Can Be Reduced (Can Be Reduced Network Delay 20-50 ms
Using LLQ) Using LFI) (Variable)
End-to-End Delay (Should be < 150 ms)

Provisioning
Voice Is Not Free—Especially on Low Speed Links—

Engineer the Network for Data, Voice, and Video
Voice Video Voice/Video Data Routing,

Control Etc.
LLQ = 33%
Sum of Traffic = 75% Reserved

Link Capacity
Link Capacity = (Min BW for Voice + Min BW for Video + Min BW for Data) / 0.75

Provisioning Tables for Voice Bearer Traffic
CODEC Sampling Rate Voice Payload Packets per Bandwidth per

in Bytes Second Conversion
G.711 20 msec 160 50 80 kbps
G.711 30 msec 240 33 74 kbps
G.729A 20 msec 20 50 24 kbps
G.729A 30 msec 30 33 18 kbps
A More Accurate Method for Provisioning Is to Include

the Layer 2 Headers into the Bandwidth Calculations:
CODEC Ethernet PPP ATM Frame Relay
14 Bytes of Header 6 Bytes of Header 53 Bytes Cells with 4 Bytes of Header
a 48 Byte Payload
G.711 at 50 pps 85.6 kbps 82.4 kbps 106 kbps 81.6 kbps
G.711 at 33 pps 77.6 kbps 75.5 kbps 84 kbps 75 kbps
G.729A at 50 pps 29.6 kbps 26.4 kbps 42.4 kbps 25.6 kbps
G.729A at 33 pps 22.2 kbps 20 kbps 28 kbps 19.5 kbps

Provisioning Tables for Signaling Traffic
Gatekeepers
GK
...
Centralized Call Processing
# of IP Phones,
Bandwidth
...
Gateways
1 to 30 8 kbps
Distributed Call Processing
50 11 kbps
# of Virtual
Bandwidth
100 23 kbps Tie Lines
150 34 kbps 1 to 70 8 kbps

What We Have Built so Far
Si
IP WAN
Si
Branch Access
Router Switch
WAN Branch
Access Distribution/ Aggregation
Switch Core Switch
Campus
Agenda
• Introduction
• Applications

Telephony Infrastructure Agenda (1/2)
• Deployment Models
• Signaling Protocols
• Gateways
• Media Resources
• Call Processing/provisioning

Deployment Models
Single Site
• Cisco CallManager,
applications and
DSP resources at same
physical location Applications
(VMail, IVR, ICD,…)
• Supports up to 30,000
CallManager
lines per cluster Cluster
• Multiple clusters can

be interconnected
via inter-cluster
trunks PSTN
• PSTN used for

all external calls

Deployment Models
Centralized Call Processing with Branches
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 Branch B
• Supports up to 30,000 lines per cluster

• Call admission control (limit number of calls per site)
• Survivable remote site telephony for remote branches
Deployment Models
Distributed Call Processing
CallManager
Applications Cluster
(VMail, IVR, ICD,…)
PSTN Applications
CallManager
Cluster
IP WAN
Branch A
GK CallManager
Headquarters Gatekeeper Cluster
• CallManager and applications Applications

located at each site
• Up to 30,000 lines per site
• 100+ sites
• Transparent use of PSTN if Branch B
IP WAN unavailable
• Gateways
• Media Resources
• Call Processing/Provisioning

Signaling Protocols
CallManager as a “Protocol Translator”
MGCP Call Agent

Gateway
Applications
Server
Analog
Phone
MGCP CTI / QBE PC-Based
IP Phone
SCCP H.323
Sound Station H.323
Gateway
IP Feature SIP
Phone (Future)
Analog
Wireless IP Phone Phone
Analog Feature
Phone Video
PC-based Terminal
IP Phone
Signaling Protocols
More about SIP
IOS Cisco SIP BTS PGW

SIP-Based ATA Proxy Server Unity
IP Phone 186 Gateways 10200 2200
Telephony
Infrastructure
Si
Network
IP WAN Infrastructure
Si
• Several Cisco voice products already support SIP

• The network infrastructure is independent of the signaling protocol
• Many PBX features cannot be delivered natively using SIP today
• Easy migration of current solution when SIP catches on in the
enterprise and rolls into CallManager
• Gateways
• Media Resources

Gateways
Gateway Selection Criteria
CallManager
PSTN
Router/ IP
Gateway
WAN
• Voice port density requirements
• Signaling protocol (H.323, MGCP, etc.)
• Support for required PSTN signaling types
• Support for required WAN interfaces and QoS
Gateways
H.323
TDM IP
PSTN
PRI Layer 3
Layer 2 H.225
Framing
Cisco CallManager
• All PSTN signaling terminates on gateway

• H.225 communication between gateway and CallManager
• H.323 is a “peer-to-peer” protocol

Gateways
H.323—Pros and Cons
Pros Cons
• Interoperability • Higher
administration
• Breadth of product required
and interface choice
• No call
• Support for preservation
survivable remote
site telephony
• Gateway intelligence

Gateways
MGCP—PRI Backhaul
TDM IP
PSTN
PRI Layer 3 PRI Layer

Layer 2
Framing MGCP
Call Signaling Cisco CallManager
• Framing and layer 2 signaling terminates at

the gateway
• Layer 3 signaling is backhauled to the
CallManager
• MGCP is a “client-server” protocol
Gateways
MGCP—Pros and Cons
Pros Cons
• Ease of dial plan • Dependency on
administration connectivity to
call agent
• Call preservation
• Port-level control
(Required for voice
mail integration)

• Gateways
• Media Resources

Media Resources
Conferencing, Transcoding, Music on Hold
IVR Call
• Conferencing Music
on Hold
Manager
Cluster
Server
DSPs needed for multi-
party conferences Transcoding
DSPs Xcode
• Transcoding Conferencing
DSPs Conf
Multiple CODEC support
(e.g., G.711 to G.729)
Automatic CODEC selection PSTN IP WAN
DSPs needed in presence
of single-CODEC endpoints
• Music on Hold
Centralized server sends
...
streams across the WAN
• Basic Call Processing
• Gateways
• Media Resources

CallManager Redundancy and Scalability
What Is a Server? What Does It Do?
Gateway
Devices Management Interface
“Register” with
M Directory Services
Transcoding CallManager
A
J Music on Hold
Xcode
O
Conferencing
R Software Conferencing
Conf Software MTP
IP Phone P
R Call Processing
O
Config Server
JTAPI C
IP-IVR CallManager E
Server CTI/QBE Interface
S
S SCCP Interface
Voice Mail E
Server S MGCP Interface
H.323 Interface
Clustering—1:1 Redundancy (CM 3.3 and MCS 7845)
Primary/Backup • Distribute IP phones based on DN

• Load-share between
Backup/Primary primary and backup servers
To 7,500 IP Phones To 15,000 IP Phones To 30,000 IP Phones

(10,000 Device Units) (20,000 Device Units) (40,000 Device Units)
Publisher and
Publisher and Publisher and TFTP Server(s)
TFTP Server(s) TFTP Server(s)
3751 to 1 to
7500 3750
3751 to 1 to
1 to 3750: Primary 11,251- 7501-
3751 to 7500: Backup 7500 3750
15,000 11,250
3751 to 7500: Primary 11,251- 7501-
1 to 3750: Backup 18,251- 15,001-
15,000 11,250 22,500 18,250
26,251- 22,501-
30,000 26,250

Clustering over the WAN—Guidelines
< 40ms RTD

Publisher/
TFTP
QoS-Enabled BW
• Max 40ms round-trip delay between ANY two

CallManagers
• 900 kbps for each 10,000 BHCA within the cluster
• Four active locations maximum (4 active CMs)
• Failover across the WAN supported (Additional BW)
Check Out the IP Telephony Design Guide for CallManager 3.1 and 3.2 for Full Details

CallManager Provisioning
Analogy PBX—IP Telephony
Gateway Conferencing Transcoding
Conf Xcode
Voice Mail
PBX Server
JTAPI
IP-IVR
T1
IP Phone
Trunk Station CallManager PC-based

Card Card IP Phone
• “Max. 500 lines” • “Max. 2,500 IP phones”

• N slots available • 5,000 device units available
• Actual number of phones • Actual number of IP phones
depends on mix of depends on mix of registered
station/trunk cards devices
CallManager Device Weights Table
Weight Weight Weight Weight

BHCA < 6 BHCA < 12 BHCA < 18 BHCA < 24
CTI Server Port 2 4 6 8
CTI Client Port 2 4 6 8
CTI 3rd Party 3 6 9 12
CTI Agent 6 12 18 24
CTI Route Point 2 4 6 8
SCCP Client 1 2 3 4
H.323 Client 3 6 9 12
H.323 Gateway 3 N/A N/A N/A
Transcoder MTP 3 N/A N/A N/A
MGCP GW 3 N/A N/A N/A
Conference 3 N/A N/A N/A
Note: GW’s, xcoder/mtp and Conference BHCA = Busy Hour Call Attempts
Bridge Are Per DSO or Session
CallManager Server Platforms
Platform Device Units Maximum IP

Per Server Phones Per Server
MCS 7845 (Dual) 10000 7500
MCS 7835 All Models 5000 2500
Compaq DL380 5000 2500
IBM xSeries 342 5000 2500
MCS 7825 2000* 1000*
SPE310 (ICS 7750) 2000* 1000*
Compaq DL 320 2000* 1000*
IBM xSeries 330 2000* 1000*
MCS 7815-1000 400 200
* The Maximum Number of IP Phones Supported on a Single Non-HA Platform Is 500;

with a Redundant Server Configuration this Caveat Is Eliminated
• Call Admission Control

• Survivable Remote Site Telephony
• Dial Plan
• Security
• Management

Call Admission Control
Why Is It Needed?
Circuit-Switched Packet-Switched Networks
Networks
IP WAN IP WAN Link Provisioned

PSTN for 2 VoIP Calls (Equivalent
to 2 “Virtual” Trunks)
IP WAN
Physical Link
Trunks
3rd Call Call No Physical
Rejected Router/
Gateway Manager Limitation on IP Links
PBX STOP
If 3rd Call Accepted,
Voice Quality of All
Calls Degrades
Call Adm. Control Limits # of VoIP Calls on Each WAN Link

CallManager “Locations”
Central
• Prevent WAN link over- Site
subscription by limiting
<Null>
voice bandwidth
Location
• Assign bandwidth limit
for voice per location
• When resources are 3
insufficient, phone gets PSTN IP WAN
fast-busy tone and a
message is displayed 2 Remote
STOP
1 Sites
• If AAR is enabled, the
call is automatically
rerouted across the
Location 1 Location 2
PSTN (requires CM 3.3) Max BW = 24 kbps Max BW = 48 kbps

Gatekeeper
San GK New
Jose Backup 1 1 Backup York
HQ Gatekeeper
2 for CAC and 2
Publisher Dial Plan Resolution Publisher
Backup 3 3 Backup
4 4
• Gatekeeper provides call admission control in presence

of multiple CallManager clusters (distributed call
processing deployments)
• Configure CallManager with “anonymous device” (CM 3.2)
or “GK-controlled inter-cluster trunk” (CM 3.3) to use
gatekeeper also to resolve E.164 addresses

• Dial Plan
• Security
• Management

Survivable Remote Site Telephony
(SRST) Mode of Operation
Normal
WAN Operation
Failure CallManager
Cluster
ISDN Backup Data
Signaling Traffic Traffic

Signaling Traffic
IP WAN
SRST
Router Voice Traffic Central Site
Remote Site
PSTN
Voice Traffic
• SRST router needs minimal configuration
• Subset of features available to the phones
(DID, DOD, Call Hold, Transfer, Speed Dial, Caller ID)
• ISDN backup may be used for data traffic—
ACL needed on branch router to block signaling traffic

• Dial Plan
• Security
• Management

Dial Plan
The “IP Routing” of IP Telephony
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
Dial Plan
CallManager Call Routing Logic
CallManager Call Routing Logic
Route Patterns
User Dials
“1200” 1XXX
12XX
User Dials Directory Numbers
“1234” 1234
1234
• CallManager matches the most specific pattern

(longest-match logic)
• An IP phone directory number is a special case
of route pattern that matches a single number
Defining External Routes
External Route Elements in CallManager
Route Pattern Route

• Matches dialed number for external calls Pattern
• Performs digit manipulation (optional)
• Points to a 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
Devices IP WAN PSTN

• Gateways (H.323, MGCP)
• Gatekeeper
• Inter-Cluster Trunk (remote CM)

Building Classes of Service
Routing by User Class or Location
Call
CallManager Manager
Cluster
Central
International Calls Site
Exec Phones
Local Calls PSTN IP WAN

Office Phones Remote
Sites
Lobby Phones
Emergency
Calls ...
Create “Dial Plan Policy Groups” Instruct these Phones to Use Their
to Define Calling Restrictions Local Gateway for PSTN Access
Example of a Dial Plan
Calling Search Route Route
Partitions Lists Groups Devices
Spaces
Calling
Search
Space Internal
assigned All IP Phones
to IP Phone
based on 911 Route
policy Internal Only Patterns
9.911
Local
Local
9.[2-9]XXXXXX
PSTN PSTN PSTN
RL RG
National
National 9.1 [2-9]XX
[2-9]XXX XXXX
International International
9.011!
9.011!#
Another Example of a Dial Plan
Calling Calling Search Route Route
Search Partitions Lists Groups Devices
Space Spaces
assigned
to IP Phone SD2911 Route
based on 911 SD 2
policy 9.911
Patterns
SD2Internal Gateways
SD2 SD2
SD2Local PSTN PSTN PSTN
9.[2-9]XXXXXX
SD2Local
2nd Choice
SD2International SD2
SD 2 9.@ IPWAN 1st Choice
SD2Internat’l
Phones IP
IPWAN WAN
OnNet GK
Anonymous
LA911 1st Choice
911 Device
LAInternal LA
9.911
IPWAN 2nd Choice
LALocal LALocal
9.[2-9]XXXXXX LA LA
PSTN PSTN PSTN
L.A.
Phones LAInternat’l LAInternational LA
9.@ Gateways

• Dial Plan
• Security
• Management

Security
Areas of Focus
Server and Apps A Endpoints
• OS updates
• Disable unused • Use separate
applications and addressing for
services voice and data
• Toll fraud • RFC1918 is
prevention preferred
measures
Firewalls
Network
• Allow only
required
applications • Secure access
• Control source (TACACS+, SSH,
addresses RADIUS)
• Use VLANs
Perimeter • Use IP filters
between voice
• No NAT across and data network
Internet
• IOS DoS tools Internet IP WAN PSTN
• Use sensors

• Dial Plan
• Security
• Management

Management
Provisioning and Reporting Element Management
• CW2000: RME 3.2,

• Provisioning: BAT Campus Mgr 3.1
(IP phones), CVM
• In-line powered
(network), QPM
switches, CCM
PRO (QoS)
• Handset tracking
• Reporting:
CAR • CCM topology
display
Fault Detection Performance Analysis
• IP Telephony
• IPM 2.2
Manager (ITEM)
• Real-time data on
• Pro-active fault
delay, jitter,…
detection
• Generate alarms
• Real-time status
based on perf.
reports on CCM,
thresholds
GWs, Apps

Gateway/ Telephony
Remote
Xcode
PSTN
Conf
Si
IP WAN
Si Branch Access
Router Switch Branch
Access Distribution/ WAN
Switch Aggregation
Core Switch
Campus
Agenda
• Introduction
• Applications

Application’s Layer
• LDAP/Directory Integration
• TAPI/JTAPI
• XML/Phone Services
• SCCP Based Applications

LDAP Directories
Directory Access and Integration
Corporate
LDAP Directory
Access Integration
(User Search)
Web
Server
Endpoints CallManager Cluster(s) Applications Voice Mail/ UM

(IP Phones, SoftPhone) (IP-IVR, ICD, PA) (Unity)
• Directory access: Endpoints enabled to search

corporate directory
• Directory integration: User profile stored in a single
repository—Single point of user authentication
LDAP Directories
Directory Access for IP Phones
Corporate
2. LDAP Search LDAP
Directory
Web Server No Integration Needed

1. HTTP
Request
3. HTTP Response (XML)
Cisco Embedded
CallManager LDAP
Directory

LDAP Directories
Directory Integration
CallManager
Cluster 1 = CallManager-Specific
Information
User
Authentication
LDAP
Corporate
LDAP Directory
User CallManager
Preferences Cluster 2
• Use the corporate LDAP directory instead of the embedded

directory to store application-specific user information
• Supported directories: Microsoft active directory, netscape
directory server
(J)TAPI and CTI Concepts
Functional Blocks
Application CTI RP CTI Port
(J) TAPI CTI Manager CallManager
RTP CTI/QBE CTI/QBE SDL SDL
Application CTI Manager CallManager

Uses the TAPI TSP Acts as a Broker Controls the CTI
or JTAPI Plug-in to between the Devices and Provides
Communicate with Application and the Device Redundancy
the CTI Manager CallManager Cluster and Call Control
Using CTI/QBE
over TCP/IP
TAPI = Telephony Application Programming Interface QBE = Quick Buffer Encoding

TSP = Telephony Service Provider SDL = Specification and Description Language
CTI = Computer Telephony Integration
© 2003, Cisco Systems, Inc. All rights reserved.
RP = Route Point 71
(J)TAPI and CTI Concepts
Joining All the Elements
1. Incoming Call from GW

RP to IVR’s Route Point
1 (DN 1000) (Pilot Number 1000)
IVR Application Redirect to
a Free IVR CTI Port (1001)
CTI Port
2 (DN 1001) IVR
2. IVR Answers Call on CTI
Port; Plays Messages and
CTI Port Collects Information,
(DN 1002) Blind Transfers Caller to
ICD Route Point (2000)
4
3
3. ICD Receives Incoming
Call from IVR, Checks to
ICD
RP
3rd Party Control (DN 2000) Agent Is On-Hook via
rd
3 Party Control; Redirects
Calls to Agent (3000)
Agent
(DN 3000)
IVR = Interactive Voice Response
ICD = Intelligent Call Distribution
Voice Mail/ Auto-Attendant/ XML LDAP
Unified
Messaging
Interactive Phone
Services
Directory Applications
Voice Response

Gateway/ Telephony
Remote
Xcode
PSTN
Conf
Si
IP WAN
Si Branch Access
WAN Router Switch Branch
Access Distribution/
Aggregation
Campus
Switch Core Switch Router Network Infrastructure
Summary Conclusions
• What are the key • IP Telephony is

components and mainstream technology
requirements of an IP
telephony solution • Key advantages are
cost, flexibility and
• How to build it: applications
Network infrastructure
Telephony infrastructure
• To learn more about IP
Telephony design:
Applications
• What are the design
guidelines and
recommendations
http://www.cisco.com/go/srnd/
Recommended Reading
Cisco CallManager
Fundamentals
ISBN: 1587050080
Voice-Enabling the Data

Network: H.323, MGCP, SIP,
QoS, SLAs, and Security
ISBN: 1587050145
Voice over IP Fundamentals

ISBN: 1578701686
Available on-site at the Cisco Company Store

Recommended Reading
Cisco IP Telephony
ISBN: 1587050501
Cisco Voice over Frame

Relay, ATM, and IP
ISBN: 1578702275
IP Telephony Unveiled
ISBN: 1587200759
Available on-site at the Cisco Company Store

PWR-5004
8026_05_2003_c3 © 2003, Cisco Systems, Inc. All rights reserved. 77

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Designing an Enterprise

© 2003, Cisco Systems, Inc. All rights reserved. 1

© 2003, Cisco Systems, Inc. All rights reserved. 2

End-to-End IP Telephony with Applications

© 2003, Cisco Systems, Inc. All rights reserved. 3

• Understanding what can be built today

© 2003, Cisco Systems, Inc. All rights reserved. 4

Telephony Media Call Processing PSTN Telephony Infrastructure

© 2003, Cisco Systems, Inc. All rights reserved. 7

• Building a Campus Network

© 2003, Cisco Systems, Inc. All rights reserved. 8

2 3 4 5 6 7 8 9 Unique VLAN 20 21 22 23 etc.

• Create a single VLAN for

• Layer 2 roaming within

Distribution HSRP HSRP

• HSRP, HSRP VVID=110 VVID=111 VVID=310 VVID=311 VVID=312

Path costs Model #1 Model #2

Core Layer Features:

by default VVID=110 VVID=111 VVID=310 VVID=311 VVID=312

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Scheduling: Assign Packets to One of Multiple Queues (Based on

Provisioning: Accurately Calculate the Required Bandwidth

Campus Branch Office

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Bi-Directional QoS Bi-Directional QoS

• Downstream traffic from AP receives “soft QoS” as

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• G.711—7 concurrent calls

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Branch 1 Branch 2 Branch 3 Branch 1 Branch 2 Branch 3

Hub and Spoke Full Mesh

Campus Branch Office

Variable Variable 6.3 µs/Km +

End-to-End Delay (Should be < 150 ms)

Voice Is Not Free—Especially on Low Speed Links—

Voice Video Voice/Video Data Routing,

Sum of Traffic = 75% Reserved

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CODEC Sampling Rate Voice Payload Packets per Bandwidth per

A More Accurate Method for Provisioning Is to Include

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• Multiple clusters can

• PSTN used for

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• CallManager at central site

• Supports up to 30,000 lines per cluster

• CallManager and applications Applications

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MGCP Call Agent

IOS Cisco SIP BTS PGW

• Several Cisco voice products already support SIP

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• All PSTN signaling terminates on gateway

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PRI Layer 3 PRI Layer

Call Signaling Cisco CallManager

• Framing and layer 2 signaling terminates at

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