Cisco Systems. Cisco Voice over IP. Volume 2. Student Guide

310 Pages·2.771 MB·English

CIPT1
Cisco Voice over IP
Volume 2
Version 6.0

Student Guide

Editorial, Production, and Web Services: 02.15.08

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Table of Contents
Volume 2
VoIP Gateway Implementation 3-1
Overview 3-1
Module Objectives 3-1
Implementing H.323 Gateways 3-3
Overview 3-3
Objectives 3-3
H.323 Gateway 3-4
H.323 and IP 3-5
H.323 Adapted to an IP Example 3-6
Why H.323 3-8
Regional Requirements Example 3-10
H.323 Network Components 3-12
H.323 Terminals 3-12
H.324 Terminals 3-13
H.323 Gateways 3-14
Cisco UBEs 3-15
H.323 Gatekeepers 3-17
Multipoint Control Units 3-19
H.323 Call Establishment and Maintenance 3-20
H.323 Call Signaling 3-21
Basic Call Setup 3-21
H.323 Fast Connect Call Setup 3-22
H.323 Multipoint Conferences 3-23
Configuring H.323 Gateways 3-25
H.323 Configuration Example 3-26
Configuring Codecs on an H.323 Gateway 3-28
Tuning H.323 Timers 3-31
Configuring H.323 Fax Pass-Through and Relay 3-33
Fax Pass-Through Example 3-33
Fax Relay Example 3-35
Configuring H.323 DTMF Relay 3-37
Verifying H.323 Gateway Status 3-39
Summary 3-41
Lesson Self-Check 3-42
Lesson Self-Check Answer Key 3-45
Implementing MGCP Gateways 3-47
Overview 3-47
Objectives 3-47
MGCP Components 3-48
Why MGCP 3-50
MGCP Architecture 3-52
MGCP Gateways 3-54
MGCP Call Agents 3-57
Basic MGCP Concepts 3-58
MGCP Calls and Connections 3-59
MGCP Control Commands 3-60
Package Types 3-62
MGCP Call Flows 3-63
Configuring MGCP Gateways 3-65
MGCP Residential Gateway Configuration Example 3-66
Configuring an MGCP Trunk Gateway Example 3-68
Configuring Fax Relay with MGCP Gateways Example 3-69
Verifying MGCP 3-71
Summary 3-76
Lesson Self-Check 3-77
Lesson Self-Check Answer Key 3-80
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for the sole use by Cisco employees for personal study. The files or printed representations may not be
used in commercial training, and may not be distributed for purposes other than individual self-study.
Implementing SIP Gateways 3-81
Overview 3-81
Objectives 3-81
SIP Fundamentals 3-82
How SIP Works 3-85
Why SIP 3-87
SIP Architecture 3-88
SIP Call Flow 3-90
Direct Call Setup 3-90
Call Setup Using a Proxy Server 3-91
Call Setup Using a Redirect Server 3-93
SIP Addressing 3-94
SIP Addressing Variants Example 3-94
SIP DMTF Considerations 3-97
Configuring SIP 3-100
Configuring a SIP Gateway Example 3-101
SIP Dial Peer Example 3-105
Verifying SIP Gateways 3-106
Summary 3-115
Lesson Self-Check 3-116
Lesson Self-Check Answer Key 3-119
Module Summary 3-121
References 3-121
Dial Plan Implementation on Voice Gateways 4-1
Overview 4-1
Module Objectives 4-1
Understanding Dial Plans 4-3
Overview 4-3
Objectives 4-3
Defining Dial Plans 4-4
Planning Considerations 4-7
Endpoint Addressing 4-9
Call Routing and Path Selection 4-10
Digit Manipulation 4-11
Calling Privileges 4-12
Call Coverage 4-13
Scalable Dial Plans 4-14
PSTN Dial Plan Requirements 4-18
Inbound PSTN Calls Example 4-19
Outbound PSTN Call Example 4-20
ISDN Dial Plan Requirements 4-21
Inbound ISDN Call Example 4-22
Configuring PSTN Dial Plans 4-23
PSTN Dial Plan Example 4-24
Complete Configurations 4-31
Inbound PSTN Call Flow Example 4-32
Outbound PSTN Call Flow Example 4-33
Verifying PSTN Dial Plans 4-34
Q.850 Cause Codes 4-35
debug isdn q931 4-40
Summary 4-44
Lesson Self-Check 4-45
Lesson Self-Check Answer Key 4-47
Implementing Numbering Plans 4-49
Overview 4-49
Objectives 4-49
Numbering Plan Characteristics 4-50
ii Cisco Voice over IP (CVOICE) v6.0 © 2008 Cisco Systems, Inc.
The PDF files and any printed representation for this material are the property of Cisco Systems, Inc.,
for the sole use by Cisco employees for personal study. The files or printed representations may not be
used in commercial training, and may not be distributed for purposes other than individual self-study.
Numbering Plan Categories 4-51
Private Numbering Plan Design Considerations 4-52
PSTN Numbering Plan Common Elements 4-52
NANP Example 4-53
Scalable Numbering Plans 4-54
Overlapping Numbering Plans 4-55
Private and Public Numbering Plan Integration 4-56
Integration of Internal and Public Numbering Plans Example 4-59
Enhancing and Extending an Existing Plan to Accommodate VoIP 4-60
Number Normalization Example 4-60
911 Services 4-62
911 Call Processing 4-64
Implementing a Numbering Plan Example 4-65
Summary 4-67
Lesson Self-Check 4-68
Lesson Self-Check Answer Key 4-70
Configuring Digit Manipulation 4-71
Overview 4-71
Objectives 4-71
Digit Manipulation 4-72
Digit Collection and Consumption 4-75
Digit Collection Example 4-76
Digit Stripping 4-77
Digit Forwarding 4-78
Digit Prefixing 4-79
Number Expansion 4-80
Simple Digit Manipulation for POTS Dial Peers Example 4-81
Number Expansion Example 4-82
Caller ID Number Manipulation 4-83
CLID Commands 4-83
Station ID Commands 4-84
Voice Translation Rules and Profiles 4-88
Understanding Regular Expressions in Translation Rules 4-92
Search and Replace with Voice Translation Rules Example 4-94
Voice Translation Profiles 4-96
Translation Profile Processing 4-98
Voice Translation Profile Search-and-Replace Example 4-99
Voice Translation Profile Call Blocking Example 4-101
Voice Translation Profiles vs. dialplan-pattern 4-103
Cisco Unified Communications Manager Express with dialplan-pattern Example 4-104
Cisco Unified Communications Manager Express with Voice Translation Profiles Example 4-105
Verifying Voice Translation Rules 4-106
Configuring Digit Manipulation 4-108
Basic Digit Manipulation Example 4-108
Voice Translation Rule and Profile Example 4-110
Summary 4-112
Lesson Self-Check 4-113
Lesson Self-Check Answer Key 4-115
Configuring Path Selection 4-117
Overview 4-117
Objectives 4-117
Call Routing and Path Selection 4-118
Dial Peer Matching 4-119
Matching to Inbound and Outbound Dial Peers 4-122
Dial-Peer Call Routing and Path Selection Commands 4-124
Matching Dial Peers in a Hunt Group 4-127
Best Practices 4-129
Path Selection Strategies 4-130
Site-Code Dialing and Toll Bypass 4-131
© 2008 Cisco Systems, Inc. Cisco Voice over IP (CVOICE) v6.0 iii
The PDF files and any printed representation for this material are the property of Cisco Systems, Inc.,
for the sole use by Cisco employees for personal study. The files or printed representations may not be
used in commercial training, and may not be distributed for purposes other than individual self-study.
Toll Bypass Example 4-132
Site-Code Dialing and Toll Bypass Example 4-133
Tail-End Hop-Off 4-134
TEHO Scenario Example 4-135
Configuring Site-Code Dialing and Toll Bypass 4-136
Site-code Dialing and Toll Bypass Scenario 4-136
Outbound Site-Code Dialing Example 4-142
Inbound Site-Code Dialing Example 4-143
Configuring TEHO 4-144
Complete Configuration 4-149
Summary 4-150
Lesson Self-Check 4-151
Lesson Self-Check Answer Key 4-152
Implementing Calling Privileges on Cisco IOS Gateways 4-153
Overview 4-153
Objectives 4-153
Calling Privileges 4-154
Understanding COR on Cisco IOS Gateways 4-156
COR Behavior Example 4-158
COR Example 4-161
Understanding COR for Cisco Unified SRST and Cisco Unified Communications Manager
Express 4-163
COR for SRST and Cisco Unified Communications Manager Example 4-164
Configuring COR 4-165
Configuring COR Example 4-167
Configuring COR for Cisco Unified SRST Example 4-175
Verifying COR 4-176
Summary 4-177
Lesson Self-Check 4-178
Lesson Self-Check Answer Key 4-179
Module Summary 4-181
References 4-181

iv Cisco Voice over IP (CVOICE) v6.0 © 2008 Cisco Systems, Inc.
The PDF files and any printed representation for this material are the property of Cisco Systems, Inc.,
for the sole use by Cisco employees for personal study. The files or printed representations may not be
used in commercial training, and may not be distributed for purposes other than individual self-study.
Module 3
VoIP Gateway Implementation
Overview
To provide voice communication over an IP network, Real-Time Transport Protocol (RTP)
sessions are created. These sessions are dynamically created and facilitated by one of several
call control procedures. Typically, these procedures also include mechanisms for signaling
events during voice calls, and for managing and collecting statistics about the voice calls.
There are several types of protocols used within Cisco Unified Communications networks. In
order to select the best protocol for a specific environment, network engineers need to know all
of these protocols, how they can be used, and their advantages and disadvantages.
This module focuses on three protocols that are used in Cisco Unified Communications
networks to implement gateways and offer call-control support for VoIP: the H.323 suite of
protocols, the Media Gateway Control Protocol (MGCP), and the session initiation protocol
(SIP).

Module Objectives
Upon completing this module, you will be able to describe the basic signaling protocols that are
used on voice gateways and configure a gateway to support calls using the various signaling
protocols. This ability includes being able to meet these objectives:
(cid:132) Describe the H.323 protocol stack and how to implement H.323 on gateways
(cid:132) Describe the MGCP stack and how to implement MGCP on gateways
(cid:132) Describe SIP enterprise features and how to implement SIP on gateways

The PDF files and any printed representation for this material are the property of Cisco Systems, Inc.,
for the sole use by Cisco employees for personal study. The files or printed representations may not be
used in commercial training, and may not be distributed for purposes other than individual self-study.
3-2 VoIP Gateway Implementation © 2008 Cisco Systems, Inc.
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for the sole use by Cisco employees for personal study. The files or printed representations may not be
used in commercial training, and may not be distributed for purposes other than individual self-study.
Lesson 1
Implementing H.323 Gateways
Overview
H.323 gateways are among the most common Cisco IOS voice gateways within Cisco Unified
Communications Manager environments. H.323 gateways are the endpoints on a LAN that
provide real-time, two-way communications between H.323 terminals on the LAN and other
ITU-T terminals on the network. H.323 gateways can also communicate with other H.323
gateways. Gateways enable H.323 terminals to communicate with terminals that are not H.323
terminals by converting protocols. Gateways are the point where a circuit-switched call is
encoded and repackaged into IP packets. Because gateways function as H.323 endpoints, they
provide admission control, address lookup and translation, and accounting services.
Objectives
Upon completing this lesson, you will be able to describe the H.323 protocol stack and how to
implement H.323 on gateways. This ability includes being able to meet these objectives:
(cid:132) Describe the functions that are performed by a typical H.323 gateway
(cid:132) Describe the advantages of H.323 as a voice gateway protocol
(cid:132) Describe the functional components that make up an H.323 environment
(cid:132) Describe the H.323 call establishment and maintenance process
(cid:132) Describe H.323 call signaling
(cid:132) Describe the types of multipoint conferences that are supported by H.323
(cid:132) Describe how to configure an H.323 gateway
(cid:132) Describe how to configure a single codec or codec negotiation on an H.323 gateway
(cid:132) Describe how to tune some H.323 timers
(cid:132) Describe how to configure fax pass-through and relay on H.323 gateways
(cid:132) Describe how to configure H.323 DTMF relay on an H.323 gateway
(cid:132) Describe how to verify the status of an H.323 gateway
The PDF files and any printed representation for this material are the property of Cisco Systems, Inc.,
for the sole use by Cisco employees for personal study. The files or printed representations may not be
used in commercial training, and may not be distributed for purposes other than individual self-study.
H.323 Gateway
This topic describes the functions performed by a typical H.323 gateway.
H.323 Gateway
H.323 gateways perform these services:
(cid:131) Translation between audio, video, and data formats
(cid:131) Conversion between call setup signals and procedures
(cid:131) Conversion between communication control signals and
procedures

©2008 Cisco Systems, Inc. All rights reserved. CVOICE v6.0—3-2
H.323 gateways are based on ITU-T Recommendation H.323. This ITU-T recommendation
pertains to H.323 packet-based multimedia communications systems. It describes an
infrastructure of terminals, common control components, services, and protocols that are used
for multimedia (voice, video, and data) communications.
An H.323 gateway is an optional type of endpoint that provides interoperability between H.323
endpoints and endpoints located on a Switched Circuit Network (SCN), such as the public
switched telephone network (PSTN) or an enterprise voice network. Ideally, the gateway is
transparent to both the H.323 endpoint and the SCN-based endpoint.
3-4 Cisco Voice over IP (CVOICE) v6.0 © 2008 Cisco Systems, Inc.
The PDF files and any printed representation for this material are the property of Cisco Systems, Inc.,
for the sole use by Cisco employees for personal study. The files or printed representations may not be
used in commercial training, and may not be distributed for purposes other than individual self-study.