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Study Guide for the Cisco CCIE Voice Written Exam
Author: Daryl P. Smith
Copyright© 2010 Network Learning, Inc.
Published by:
Network Learning Inc. (Cisco Learning Partner)
375 N. Stephanie Street, Building 21, Suite 2111
Henderson, NV 89014 USA
All rights reserved. No part of this book may be reproduced or transmitted in any form or
by any means, electronic or mechanical, including photocopying, recording, or by any
information storage and retrieval system without written permission from the publisher
except for the inclusion of brief quotations in a review.
Printed in the United States of America
Warning and Disclaimer
This book is designed to provide information to the Cisco CCIE Voice written exam. Every
effort has been made to make this book as complete and as accurate as possible, but no
warranty or fitness is implied.
The information is provided on an “as is” basis. The authors, editors, and Network Learning
Inc., shall have neither liability nor responsibility to any person or entity with respect to any
loss or damages arising from the information contained in this book or from the use of the
discs or programs that may accompany it.
The opinions expressed in this book belong to the author and are not necessarily those of
Network Learning Inc.
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Trademark Acknowledgements
All terms mentioned in this book that are known to be trademarks or service marks have
been appropriately capitalized. Network Learning Inc. cannot attest to the accuracy of this
information. Use of a term in this book should not be regarded as affecting the validity of
any trademark or service mark.
Author – Daryl P. Smith
Daryl Smith is the author of CCBOOTCAMP's CCIE Voice Exam Strategy Guide
Workbook. He has attained the highest level Cisco® certification: Cisco Certified
Internetwork Expert: CCIE#25893 in Voice. Daryl is an experienced engineer with over
15 years in the networking and voice arena. Currently Daryl is working toward attaining
his CCDE certification. He holds the following certifications: Cisco Certified Network
Associate (CCNA), Cisco Certified Voice Professional (CCVP), Cisco Certified Internetwork
Expert (CCIE-Voice#25893), and Cisco Certified System Instructor (CCSI) 33397.
Contributing Author – Brad Ellis
Brad Ellis (CCIE #5796, CCSI #30482, CSS1, CCDP, CCNP, MCNE, and MCSE) works as
a network engineer and is the CEO of Network Learning Inc. He has been dedicated to
the networking industry for over 12 years. Brad has worked on large scale security
assessments and infrastructure projects. He is currently focusing his efforts in the
security and voice fields. Brad is a dual CCIE (R&S / Security) #5796.
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Contents
Contents 3
Introduction 27
Chapter 1 Infrastructure Protocols 28
DNS 28
Using DNS with CUCM Advantages 28
IP Address advantages 28
Cisco TFTP 29
TFTP Process Overview for SCCP Devices 30
TFTP Process Overview for Cisco Unified IP Phones Using SIP 31
Understanding How Devices Use DHCP and Cisco TFTP 32
Obtaining an IP Address 33
Requesting the Configuration File 33
Contacting Cisco Unified Communications Manager 34
NTP 35
SCCP Phones 36
SIP Phones 36
NTP for Network Devices 37
Power over Ethernet 37
Cisco AVVID and Cisco IP Communications 39
Inline Power 39
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Cisco Inline Power 39
Statement of Direction 40
Voice and Data VLAN 41
Dynamic Host Configuration Protocol (DHCP) 41
Cisco IOS DHCP Server 41
Understanding Voice VLAN 42
Cisco IP Phone Voice Traffic 43
Cisco IP Phone Data Traffic 43
Default Voice VLAN Configuration 43
Single VLAN 44
Multi VLAN 44
Trunk Port VLAN 45
Troubleshooting Infrastructure Protocols 45
NTP 45
TFTP and DHCP 47
VLAN 48
Chapter 1 Questions 49
Chapter 1 Answers 52
Telephony Protocols 53
SCCP 53
Media Gateway Control Protocol (MGCP) 53
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Call Connections for MGCP 56
PRI and BRI Backhaul 57
Configuring MGCP 58
Session Initiation Protocol (SIP) 58
SIP Entities 60
User Agent 61
Proxy Server 61
Redirect Server 61
Registrar 61
B2BUA 62
SIP Messages 62
SIP Request Messages 62
SIP Response Messages 62
SIP Addresses 63
Fully qualified domain names 63
E.164 addresses 63
Mixed addresses 63
Configuring SIP 64
SIP DTMF signaling 65
H.323 65
Terminals 67
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Gateways 67
Gatekeepers 68
Multipoint Control Unit (MCU) 70
H.323 Version 2 71
Security 71
Fast Call Setup 71
Supplementary Services 71
T.120/H.323 Integration 72
H.323 Call Scenario 72
Basic H.323 Call Setup 72
Basic Configuration of an H.323 gateway 73
Call Flow with a Gatekeeper 74
Call Flow with multiple Gatekeepers 74
Configuring H.323 75
Strengths of H.323, SIP, and MGCP 77
H.323 77
SIP 77
MGCP 77
IP Voice Signal Interconnect CUBE 78
Requires one of these Cisco IOS feature sets: 79
Cisco Unified Border Element with Gatekeeper Network Topology 79
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Real Time transport protocol (RTP) 82
Real Time Control Protocol - RTCP 83
Analog Interfaces 85
FXS 85
FXO 87
E&M 88
Digital Signaling (T1/E1) 90
CAS Systems: T1 90
Channel Associated Signaling 92
E1 Interface 92
CAS Systems: E1 93
E1 Channel Associated Signaling 94
Common Channel Signaling 94
CCS Signaling 94
ISDN 95
QSIG 96
ISDN PRI QSIG voice-signaling provides the following benefits: 96
IOS Dial-peer and Digit Manipulations 97
Dial Peer Overview 97
Call Legs 97
Destination Patterns 99
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Types of Dial-Peers 99
Digit Manipulation 100
Wildcards 101
Digit Stripping and Prefixes 102
Forwarding Digits 104
Translation Rules 105
Features: 105
Assign Translation Profiles 105
voice translation-rule Command 107
Translation Profile Configuration 108
Dial Peer Configuration 108
Troubleshooting Telephony Protocols 108
MGCP 109
MGCP and H.323 controllers 110
ISDN debug 110
Active Calls on a Router 111
SIP 113
H.323 and Gatekeeper 115
Dial-peers 116
Call Control 118
Translation Rules 119
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Chapter 2 Questions 120
Chapter 2 Answers 124
Chapter 3 Cisco Unified Communications Manager (CUCM) 125
Device Registration 125
Device Settings 126
Device Defaults Configuration 126
Phone Button Template 126
Softkey Templates 126
SIP Profile 127
Common Profile 127
Device Redundancy 128
Call Processing Redundancy 128
SRST-MGCP Fallback 129
Codec Selection 129
Dial Plan 130
Local Route Group 132
Support for + Dialing 134
Calling Party Number Transformations 136
Called Party Number Transformations 138
Globalize Call Routing Ingress and Egress 141
Constructing a Globalize Call Flow 144
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Partitions 146
Calling Search Space 146
Digit Analysis 149
Digit collection 152
Type B SIP Phones 154
SIP Dial Rules 154
Digit Manipulation 156
Benefits of “+” Dialing 157
AAR 157
Call Forward Unregistered (CFUR) 158
Tail End Hop Off (TEHO) 158
Media Resources 158
Conference bridges 159
Media Termination Point 160
Annunciator 160
Transcoder 160
Cisco RSVP Agent 162
Music on Hold 162
Unicast and Multicast MoH 163
Cisco IP Voice Media Streaming Application 164
Media resource Manager (MRM) 165
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Media Resource Group 165
Media Resource Group List 165
CUCM Applications 167
Extension Mobility (EM) 168
Unified CM Assistant 169
Unified CM Assistant Functionality and Architecture 170
Unified CM Assistant Share Lined Mode 171
Cisco Unified Mobility 172
Mobile Connect 173
Mobile Voice Access and Enterprise Feature Access 173
Remote Destination Profile Configuration 174
CUCM CTI Integrations 174
CTI Port 176
CTI Route Point 176
CUM Serviceability and OS Administration 177
Cisco OS Administration 177
CUCM Disaster Recovery 178
Troubleshooting CUCM 178
Dial Plan and Routing Issues 179
Route Partitions and Calling Search Spaces 180
Problem When Dialing a Number 182
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Intercluster Cisco Unified IP Phone Calls 184
Intercluster H.323 Communication 185
Chapter 3 Questions 196
Chapter 3 Answers 207
Chapter 4 Cisco IOS IP Telephony 208
CUCME 208
CUCME Overview 208
SRST 209
Components of Centralized Call-Processing Architecture 210
When to user MGCP Fallback 212
When to Use Basic SRST: 212
When to Use CUCME SRST 213
SRST Timing 213
CUE 214
Deployment Models 215
Standalone Office 215
Multisite Networks 216
CUCME Call Features 217
Basic Automatic Call Distribution (B-ACD) 217
Customer Contact 218
IP Telephony 218
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Rich-Media Conferencing 218
Third-Party Applications 218
Unified Communications 218
Video Telephony 219
IOS Media Resources 219
DSP Farms 219
DSP Farm Profiles 219
Conferencing 220
Transcoding 220
Media Termination Point 221
Allocation of DSP Resources Within the DSP Farm 222
Troubleshooting IOS Telephony 223
Telephony E-Phone & SIP Registration 223
Show ephone registered 224
Show voice register pool 1 224
Show telephony-service all & show voice register global 225
Chapter 4 Questions 227
Chapter 4 Answers 228
Chapter 5 - Quality of Service (QoS) 229
QoS Overview 230
Five Benefits for Implementing QoS in the Enterprise Networks 230
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How a Converged Network Behaves Without QoS 231
QoS framework 231
Call Admission Control Functionality 231
Integrated Services vs. Differentiated Services 232
Configure QoS Policy using Modular QoS CLI 234
QoS Configuration overview 234
FIFO: default, no config necessary 234
CBWFQ: 235
CQ: 235
PQ: 235
LLQ: 236
CAR: 236
WRED: 237
FRED: 237
Link Fragmentation and Interleaving (LFI) for Multilink PPP (MLP): 238
Configure and Monitor Various LFI methods and CRTP 240
Classification and Marking 243
Purposes of Classification and Marking 243
Difference Between Classification and Marking 244
Class of Service, IP Precedence and DiffServ Code Points 244
Network Based Application Recognition (NBAR) 246
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Classify and Mark Traffic 247
Congestion Management: Queuing 249
Identify and Differentiate Between IOS Queuing Techniques 250
Apply Each Queuing Technique to the Appropriate Application 253
IP RTP Priority and Low Latency Queuing (LLQ) Differences 254
Configure WFQ, CBWFQ, and LLQ 256
Congestion Avoidance 258
Explain How TCP Responds to Congestion 258
Explain Tail Drop and Global Synchronization 258
Identify and Differentiate Between: RED, WRED, FRED 259
Configure IOS Congestion Avoidance Features 260
Link Efficiency Tools 262
The Need for Link Efficiency Tools 262
Real Time Protocol Header Compression (cRTP) 264
Policing and Shaping 264
The Difference Between Policing and Shaping and How Each Relates to QoS 264
When to Apply and How to Configure Policing Mechanisms 265
Different Types of Traffic Shaping and How to Apply Them 265
Configure the Different Types of Traffic Shaping 267
Congestion-Control Mechanisms 269
Traffic Shaping Parameters 270
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Traffic Shaping Calculation 270
First-In, First-Out (FIFO) 271
Weighted Fair Queuing (WFQ) 271
Priority Queuing 272
Custom Queuing 274
Class-Based Weighted Fair Queuing 276
Packet over SONET/SDH (PoS) and IP Precedence 277
IP Precedence 277
Random Early Detection (RED) 279
Weighted Random Early Detection (WRED) 279
Weighted Round-Robin (WRR)/Queue Scheduling 280
Class of Service (CoS) 281
Shaping vs. Policing 282
Traffic Shaping 282
Committed Access Rate (CAR) 285
Network-Based Application Recognition (NBAR) 286
Configuring NBAR 287
Differentiated Services Code Point (DSCP) 288
Resource Reservation Protocol (RSVP) 290
Load Balancing 291
802.1x and QoS 292
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LAN QoS 310
Trust boundary 311
Connecting an IP Phone 312
AutoQos 313
AutoQoS VoIP 314
AutoQoS Enterprise 315
3750 QoS 315
Default Ingress QoS Configuration 316
Congestion Management and Avoidance 317
Queueing and Scheduling 318
Egress QoS Features 319
Default Configuration 320
Chapter 5 Questions 322
Chapter 5 Answers 331
Chapter 6 – Unified Messaging 332
Integration 332
Integration Capabilities 334
Integration Functionality SCCP & SIP 336
Digital Integration with Digital PIMG Units 338
DTMF Integration with Analog PIMG Units 339
Serial (SMDI, MCI, or MD-110) Integration with Analog PIMG Units 340
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Call Information 341
Integration Functionality 341
Integration Description TIMG 342
Serial Integration with TIMG Units 342
In-Band Integration with TIMG Units 343
Call Information 344
Integration Functionality 344
Deployment Models 345
Single-Site Messaging 345
Centralized Messaging 346
Distributed Messaging 347
MWI 348
MWI Format 349
SMDI integration 349
The forwarded call format is: 351
Call Handlers 352
Directory Handlers 353
Default System Call Handlers 354
Overview of Call Handler Greetings 355
Standard Greeting for Call Handlers 356
Offering One-Key Dialing During Call Handler Greetings 357
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Offering System Transfers 358
Abbreviated Extensions: Prepending Digits to Extensions That Callers Enter 358
Taking Messages 359
Transferring Calls 359
Directory Call Handlers 361
Creating a Directory Handler 361
Interview Handlers 362
VPIM 362
Interoperability with disparate systems: 363
VPIM Concepts 363
The following VPIM concepts will be explained: 363
VPIM Messages 363
VPIM Addresses 365
Messaging Similarities and Limitations 366
Audio Format Considerations 367
Troubleshooting Unified Messaging 368
Unity Connection Traces 368
Traces with Cisco Unity Express 368
MWIs Do Not Turn On or Off 369
Task List for Troubleshooting When MWIs Do Not Turn On or Off: 369
Utilities 370
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Cisco Unity Connection Grammar Statistics Tool 370
Cisco Unity Connection Serviceability 371
Cisco Unity Connection Task Management Tool 372
Cisco Voice Technology Group Subscription Tool 372
Real-Time Monitoring Tool 372
Cisco Unified Serviceability 373
Remote Database Administration Tools 374
Cisco Utilities Database Link for Informix (CUDLI) 374
Remote Port Status Monitor 374
Chapter 6 Questions 375
Chapter 6 Answers 377
Chapter 7 Integration 378
IPCC overview 378
Deployment Models 379
IPCC Express co-resident 380
Single-Server Non-HA Deployment Model 380
Multi-Server Non-HA Deployment Model 381
Two-Server HA Deployment Model 382
Four-Server HA Deployment Model 383
Six-Server HA Deployment Model 384
Ten-Server HA Deployment Model 385
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User Accounts 386
CTI Ports 386
CTI Route Points 387
Accessing Cisco Unified CCX Administration 387
Cisco Unified Communications Manager Configuration Page 388
Resource Manager-Contact Manager 388
Control Center 389
Prompt Management 390
Script Management 391
Resource Manager-Contact Manager 393
Scripting 393
Basic Script 402
Select Resource 403
Connect Step 404
Call Hold/ Call Unhold 404
Get Reporting Statistic 404
Troubleshooting Cisco Unified CCX 406
Troubleshooting Tips 406
Debugging a Script 408
Reactive or Non-Reactive Debugging 409
Trace Log Files 410
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Alarm Configuration 411
Chapter 7 Questions 413
Chapter 7 Answers 417
Chapter 8 – Presence 418
Presence Components 418
The Message Flow, Publish, and Subscribe 421
CUPS Administration 423
CUPC – Cisco Unified Personal Communicator 423
Cisco Unified Presence − Settings. 432
Users not showing up in CUPS user list 436
Troubleshooting using the Cisco Unified Personal Communicator 437
Server Health Tool 437
Audio and Video 438
Questions Chapter 8 439
Answers Chapter 8 442
Chapter 9 - UC Security 443
Security Overview 443
DHCP Snooping 444
Phone Authentication and Encryption 445
Disabling the Gratuitous ARP Setting 446
Disabling Web Access Setting 446
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Disabling the PC Voice VLAN Access Setting 447
Disabling the Setting Access Setting 447
Disabling the PC Port Setting 448
PKI Topologies in CUCM Deployments 448
Initial Download of the CTL 453
IP Phone Usage of the CTL 453
Firewalls, Gateway Security & NAT 454
Overview for Cisco IOS MGCP Gateway Encryption 454
Overview for SIP Trunk Encryption 455
NAT 456
Implementing Firewall Traversal and NAT 456
TCP/UDP Port List 457
Chapter 9 Questions 460
Chapter 9 Answers 463
Chapter 10 - Application Protocols 464
IP Multicast 464
Benefits of IP Multicast 464
IGMP and CGMP Multicast Protocols 465
Designated Querier 467
Querier router election 467
IGMP Versions 1, 2, and 3 468
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Multicast Address Allocation 470
Static Address Allocation Methods 470
Scope Relative Address Allocation 470
Dynamic Address Allocation 471
SDR—Session Directory 471
Classic PIM-SM 471
Bidirectional PIM 472
IP Multicast Routing 473
Multicast Groups 473
Rendezvous Points (Auto-RP, BSR) 475
Recommended Rendezvous Point Placement 476
Group-RP Mapping Mechanism 476
Recommendation 476
Comments on Auto-RP 477
Comments on Static RP 478
Calculating a Multicast Address 478
Protocol Independent Multicast (PIM) 479
PIM Commands 481
Reverse Path Forwarding (RPF) 481
PIM and Distance Vector Multicast Routing Protocol (DVMRP) 481
PIM-SM Mechanics (Joining, Pruning PIM State, Mroute table) 482
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PIM-SM uses these PIMv2 messages 482
PIM-DM 483
Bidirectional PIM (bidir-PIM) 484
Designated Forwarder (DF) Election 486
Bidirectional Group Tree Building 487
Packet Forwarding 488
Memory, Bandwidth, and CPU Requirements 488
Benefits and Drawbacks of PIM 489
Debugging bidir-PIM is easier than PIM-SM 489
RP Tree Delivery for All Packets 489
Bidir-PIM Partial Upgrades Not Allowed 489
Bidir-PIM Network Redundancy Not Supported 490
Bidir-PIM Nonbroadcast Multi-access Mode Not Supported 490
Bidir-PIM Traffic Forwarding Restrictions 490
Anycast RP 492
IP Multicast Terms 492
Unicast & Multicast for CUCM/CUCME 497
Video 499
Video Codec 499
Video Call Bandwidth 500
Cisco Video Telephony Advantage (CVTA) 501
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Fax and Modem 502
Overview of Fax and Fax Relay 502
Fax Relay Basics 503
Chapter 10 Questions 506
Chapter 10 Answers 509
Chapter 11 Operation and Network Management 510
Operating System Status and Configuration 510
Settings 510
Software Upgrades 511
UC Product Upgrade 511
Traces 512
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Introduction
This book is targeted toward the potential Cisco CCIE Voice candidates preparing for
the new CCIE Voice Exam based upon Cisco Unified Communications Manager 7.0,
Unity Connection 7.0, Cisco Unified Presence Server and Cisco Unified Contact Center
Express. The written guide is more than just a guide to assist you in passing the
written exam (350-030), but to assist you with your career as well. This guide can
also be used as a reference guide for it contains a combination of notes, white
papers and Cisco technical tutorial as well classroom material from CCBOOTCAMP.
This guide also provides some sample questions that are not directly related to
actual questions you will see on the exam but questions that will help you
understand the topics and concepts within each chapter. These questions serve as a
guide and will help you build confidence as you prepare for the CCIE Voice Written
Exam. Some of the concepts are complex and this guide will help you understand
these concepts as you prepare for the written exam as well as the CCIE Voice Lab
exam.
I also recommend you read the CUCM SRND guide found on www.cisco.com/go/srnd
and additional material found on Cisco Learning Network for CCIE candidate:
https://learningnetwork.cisco.com/community/certifications/ccie_voice
This is your first step in a long journey to becoming a CCIE. This journey is not an
easy one but it is one that you will enjoy once you succeed. Once you obtain your
CCIE you will be recognized as one of the elites in the industry, for you have
demonstrated your knowledge and capability to perform complex concepts. Good
luck in your journey and remember to enjoy it.
Daryl P. Smith, September 2010
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Chapter 1 Infrastructure Protocols
DNS
DNS within Cisco Unified Communications Manager clusters has some advantages and
disadvantages. DNS is used to for name resolution and it allows CUCM services and
applications to reference the server/system by name instead of by IP address.
Cisco Unified Communications Manager can use either IP addresses or host names to refer
to other devices such as server or application settings within the cluster. When host names
are used, the CUCM server must be able to resolve the names. Therefore; a DNS server is
required within the environment.
Using DNS with CUCM Advantages
In the case of using DNS, management is simplified because logical names are simpler to
handle then 32-bit addresses. IF IP addresses change, there is no need to modify the
application settings, because they can still use the same names; only the DNS server
configuration needs to be modified in this case. IP addresses of CUCM server can be
translated by NAT toward IP Phones, because the IP phone configuration files do not include
the original server IP address. When the DNS requests are sent out by IP Phones, the use of
NAT for the server IP addresses is no problem, for the phones are using a hostname and not
the IP address.
IP Address advantages
The system doesn’t depend on DNS for name resolution. A device (IP Phone) can initiate a
request directly to the target and the time required for an established connection is
reduced. By eliminating the need for DNS, there are no errors caused by DNS
misconfiguration. Troubleshooting is simplified because there is no need to verify proper
name resolution.
Best Practice recommendation is not use DNS with Cisco Unified Communications Manager.
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If DNS is used within the environment, there are some components that will require DNS
and rely on the availability of the DNS server or servers.
IP Phones will require DNS for signaling when CUCM servers are configured by names. The
configured CUCM server is part of the configuration file of the IP Phone. Therefore an IP
Phone needs to be able to resolve the name of the CUCM server or servers to an IP address
when CUCM server is specified by name. By default CUCM servers are added to the
configuration database by their name. IP Phones also need to be capable of resolving names
of IP Addresses when IP Phone service URLs use names instead of IP addresses. This
applies to service accessed from the services button at the phone, and to services that are
accessed by phone buttons configured with service URLs.
SIP Trunks, SIP Gateways, H.323 Gateways and H.323 Trunks can be configured to use
Host Names as well as SIP route patterns and SNMP and other network management
servers.
Cisco Unified Communications Manager servers never use DNS for intracluster
communications; Cisco Unified Communications Manager servers always use IP addresses,
regardless of whether host names are configured for the servers.
Cisco TFTP
The Cisco TFTP service builds and serves files that are consistent with the Trivial File
Transfer Protocol (TFTP). Cisco TFTP builds configuration files and serves embedded
component executables, ringer files, and device configuration files. A configuration file
contains a prioritized list of Cisco Unified Communications Managers for a device (phones
that are running SCCP and phones that are running SIP and gateways), the TCP ports on
which the device connects to those Cisco Unified Communications Managers, and an
executable load identifier. Configuration files for selected devices contain locale information
and URLs for the phone buttons: messages, directories, services, and information.
Configuration files for gateways contain all their configuration information. You can find
configuration files in a .cnf, a .cnf.xml, or an .xml format, depending on the device type and
your TFTP service parameter settings. When you set the BuildCNFType service parameter to
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Build All, the TFTP server builds both .cnf.xml and .cnf format configuration files for all
devices. When you set this service parameter to Build None, the TFTP server builds
only .cnf.xml files for all devices. When this parameter is set to Build Selective, which is the
default value, the TFTP server builds .cnf.xml files for all devices and, in addition, builds .cnf
files only for a select list of device types.
TFTP Process Overview for SCCP Devices
The TFTP server can handle simultaneous requests for configuration files. The request
process is as follows: When a device boots, it queries a DHCP server for its network
configuration information. The DHCP server responds with an IP address for the device, a
subnet mask, a default gateway, a Domain Name System (DNS) server address, and a TFTP
server name or address. Cisco Unified IP Phone 796X, for example, supports up to two
TFTP servers. If the primary TFTP server is not reached, such devices attempt to reach the
fallback TFTP server. The device requests a configuration file from the TFTP server. The
TFTP server searches three internal caches, the disk, and then alternate Cisco file servers (if
specified) for the configuration file. If the TFTP server finds the configuration file, it sends it
to the device. If the configuration file provides Cisco Unified Communications Manager
names, the device resolves the name by using DNS and opens a connection to the Cisco
Unified Communications Manager. If the device does not receive an IP address or name, it
uses the TFTP server name or IP address for setting up its registration connection.
If the TFTP server cannot find the configuration file, it sends a message to the device.
"file not found"
Devices that are requesting a configuration file while the TFTP server is rebuilding
configuration files or while processing the maximum number of requests, receive a message
from the TFTP server, which causes the device to request the configuration file later. The
Maximum Serving Count service parameter, which can be configured, specifies 200 as the
maximum number of requests.
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TFTP Process Overview for Cisco Unified IP Phones Using SIP
Unlike phones that are running SCCP, phones that are running SIP get all their
configurations from the TFTP server. From initial startup, the phone that is running SIP
contacts the configured TFTP server (either manually configured or configured through the
DHCP server) to get the configuration files; it then registers itself to its configured Cisco
Unified Communications Manager.
When the configuration of the phone that is running SIP gets changed, the Cisco Unified
Communications Manager database notifies the TFTP server to rebuild all the configuration
files or to rebuild selectively. The TFTP server retrieves information from the Cisco Unified
Communications Manager database and converts it into the proper output format, according
to the device type, and saves the output in TFTP cache. When the TFTP server gets a
request, it searches either the cache or Alternate File Server locations disk to serve the
requested configuration file or default files.
The TFTP support for phones that are running SIP builds and serves different formats of SIP
configuration files from the Cisco Unified Communications Manager database for the
following Cisco Unified IP Phones:
• Cisco Unified IP Phone 7970/71, 7961, 7941, 7911 (These phones share the same
SIP configuration file format).
• Cisco Unified IP Phone 7960, 7940 (These phones share the same SIP configuration
file format).
• Cisco Unified IP Phone 7905, 7912
• SIP dial plans on the preceding phones
• Softkey templates on the preceding phones
The TFTP server generates the following files from the Cisco Unified Communications
Manager database for configuration of phones that are running SIP:
• System wide default configuration files and per-device configuration files.
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• List of system wide dial plans for Cisco Unified IP Phones 7970/71, 7960/61,
7940/41, and 7911.
• List of system wide softkey template files.
The system derives filenames from the MAC Address and Description fields in the Phone
Configuration window of Cisco Unified Communications Manager Administration and the
device name field in the Cisco Unified Communications Manager database. The MAC address
uniquely identifies the phone.
Understanding How Devices Use DHCP and Cisco TFTP
1. Cisco IP phone obtains power from the switch
2. Cisco IP phone loads locally stored image
3. Switch provides VLAN information to Cisco IP phone using Cisco Discovery Protocol
4. Phone sends DHCP request; receives IP information and TFTP server address
5. Cisco IP phone gets configuration from TFTP server
6. Cisco IP phone registers with Cisco Unified Communications Manager server, Unified
CM sends softkey template to SCCP phone using SCCP messages.
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Cisco telephony devices require IP addresses that are assigned manually or by using DHCP.
Devices also require access to a TFTP server that contains device loads and device
configuration files.
Obtaining an IP Address
If DHCP is enabled on a device, DHCP automatically assigns IP addresses to the device
when you connect it to the network. The DHCP server directs the device to a TFTP server (or
to a second TFTP server, if available for the device). For example, you can connect multiple
Cisco Unified IP Phones anywhere on the IP network, and DHCP automatically assigns IP
addresses to them and provides them with the path to the appropriate TFTP server.
If DHCP is not enabled on a device, you must assign it an IP address and configure the TFTP
server locally on the device.
The default DHCP setting varies depending on the device:
• Cisco Unified IP Phones stay DHCP-enabled by default. If you are not using DHCP,
you need to disable DHCP on the phone and manually assign it an IP address.
• DHCP always remains enabled for Cisco Access Analog and Cisco Access Digital
Gateways.
Requesting the Configuration File
After a device obtains an IP address (through DHCP or manual assignment), it requests a
configuration file from the TFTP server. If a device has been manually added into the Cisco
Unified Communications Manager database, the device accesses a configuration file that
corresponds to its device name. If a phone is not manually configured and auto-registration
is enabled, the phone requests a default configuration file from the TFTP server and starts
the auto-registration procedure with Cisco Unified Communications Manager. If a phone
has an XML-compatible load, it requests a .cnf.xml format configuration file; otherwise, it
requests a .cnf file.
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Contacting Cisco Unified Communications Manager
After obtaining the configuration file from the TFTP server, a device attempts to make a TCP
connection to the highest priority Cisco Unified Communications Manager in the list that is
specified in the configuration file. If the device was manually added to the database, Cisco
Unified Communications Manager identifies the device. If auto-registration is enabled in
Cisco Unified Communications Manager, phones that were not manually added to the
database, attempt to auto-register in the Cisco Unified Communications Manager database.
Cisco Unified Communications Manager informs devices that are using .cnf format
configuration files of their load ID. Devices that are using .xml format configuration files
receive the load ID in the configuration file. If the device load ID differs from the load ID
that is currently executing on the device, the device requests the load that is associated
with the new load ID from the TFTP server and resets itself. A phone gets the Ring Tones
list after it performs its booting process, when the user wants to modify the Default Phone
Ring setting, and when the user loads new ring tones.
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NTP
Date and Time are important for devices within a Cisco Unified Communications Cluster and
all of the applications servers used as well; Cisco Unity Connection, Cisco Unified Presence,
etc. Network devices as well as servers may require time synchronization and replication of
data within a database cluster to have specific time stamps.
Network Time Protocol (NTP) is a protocol for synchronizing the clocks of systems within an
IP network. NTP has a hierarchical organization using clock strata. Stratum 0 is an
extremely precise clock source, such as an atomic clock or a radio clock. A stratum 1 server
is directly connected to a stratum 0 clock and can provide time information to other devices,
which themselves can serve stratum 3 devices.
The use of NTP on all devices ensures that they all have synchronized clocks. The Publisher
sends NTP request to an external NTP server and the subscribers always synchronize their
time with the publisher. The configuration of an external NTP server is not mandatory. If no
NTP server is configured, the publisher relies on its own system time. NTP can be enabled
and configured during installation or after installation. To configure NTP after installation
you make the changes with Cisco Unified Communication Manager OS Administration.
Settings-> NTP Servers.
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System time is important within CUCM cluster. The following items depend upon an
accurate clock being set and synchronized:
• Cisco IP Phone display date and time information
• Call Detail Records and Call Management Record (CDR and CMR)
• Alarms and event logs and Trace Files information for Troubleshooting
• Some Cisco Unified Communications Manager Features are date or time-based and
therefore rely on having the correct date and time. Time-of-day routing and
certificate-based security features
To ensure that all network devices have the correct date and time it is recommended that
all network devices use NTP for time synchronization. The master reference clock should be
a stratum 1 NTP server.
SCCP Phones
SCCP phones obtain date and time information from the Cisco Unified Communications
Manager. The Date/Time Group value, which is configured at the phones device pool, is
considered to allow phones deployed at different time zones to display the local time only,
not sync their time.
SIP Phones
SIP phones on the other hand must obtain their time from one or more configured NTP
references. These references are added to the Date/Time group and are therefore applied to
the phone via the device pool. The NTP reference is applicable only to SIP phones. If a SIP
Phone doesn’t have a NTP reference configured, or none of the configured servers are
reachable, the SIP phone obtains time information from the SIP signaling message received
from CUCM. It extracts the time from the time stamp of the 200 OK messages it receives.
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NTP for Network Devices
For network devices with a Cisco Unified Communications cluster, IOS devices can act as
NTP servers or clients. They can transmit either unicast or broadcast messages to server or
clients.
The Cisco IOS implementation of NTP supports additional features such as authentication
and access restrictions. If a Cisco IOS device is configured as an NTP server, it can be used
as an external NTP server by the CUCM publisher. As stated in the previous section for
CUCM NTP, subscribers always synchronize their time from the publisher. An external NTP
server can be configured only at the publisher server, not on any other nodes within the
cluster.
To configure an IOS device as a NTP server, the following commands must be entered on
the device.
Router (config) #ntp master 2
! configures router as an NTP source
Router (config)#ntp source loopback 0
! configures source interface for NTP
Router (config)#ntp server 10.1.200.229
! specifies the NTP Server
You should always configure your Cisco IOS devices within your environment with a NTP
reference. Also you should configure summer time range and date and time stamps for
debugging and logging.
Power over Ethernet
Power over Ethernet (PoE) allows the LAN switching infrastructure to provide power to an
endpoint ("powered device") over a copper Ethernet cable. This capability, once referred to
as "inline power," was originally developed by Cisco in 2000 to support emerging IP
telephony deployments.
IP telephones need power for operation, and Power over Ethernet supports scalable,
manageable power delivery and simplifies IP telephony deployments. As wireless
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