Cisco Voice Gateways and Gatekeepers (Hardcover)
David Mallory, Ken Salhoff, Denise Donohue
- 出版商: Cisco Press
- 出版日期: 2006-08-27
- 售價: $2,840
- 貴賓價: 9.5 折 $2,698
- 語言: 英文
- 頁數: 648
- 裝訂: Hardcover
- ISBN: 158705258X
- ISBN-13: 9781587052583
-
相關分類:
Cisco
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Description
Cisco Voice Gateways and Gatekeepers
Understanding and configuring GW/GK in complex VoIP networks
Denise Donohue, CCIE® No. 9566
David Mallory, CCIE No. 1933
Ken Salhoff, CCIE No. 4915
Deployments of voice over IP (VoIP) networks continue at a rapid pace. Voice gateways are an essential part of VoIP networks, handling the many tasks involved in translating between transmission formats and protocols and acting as the interface between an IP telephony network and the PSTN or PBX. Gatekeepers and IP-to-IP gateways help these networks scale. Gatekeepers provide call admission control, call routing, address resolution, and bandwidth management between H.323 endpoints including Cisco IOS® voice gateways and Cisco® Unified CallManager clusters. IP-to-IP gateways allow VoIP calls to traverse disparate IP networks.
Cisco Voice Gateways and Gatekeepers provides detailed solutions to real-world problems encountered when implementing a VoIP network. This practical guide helps you understand Cisco gateways and gatekeepers and configure them properly. Gateway selection, design issues, feature configuration, and security and high-availability issues are all covered in depth. The abundant examples, screen shots, configuration snips, and case studies make this a truly practical and useful guide for anyone interested in the proper implementation of gateways and gatekeepers in a VoIP network. Emphasis is placed on the accepted best practices and common issues encountered in real-world deployments.
Cisco Voice Gateways and Gatekeepers is divided into four parts. Part I provides an overview of an IP voice network. Part II is dedicated to voice gateways, including discussions of Media Gateway Control Protocol (MGCP); H.323; Session Initiation Protocol (SIP); voice circuit options; connecting to the PSTN, PBX, and IP WAN; dial plans; digit manipulation; route selection; class of restriction; Survivable Remote Site Telephony (SRST) and MGCP fallback; digital signal processor (DSP) resources; and Tool Command Languaue (Tcl) scripts and Voice XML (VXML). Part III addresses voice gatekeepers, including detailed deployment and configuration. Part IV is dedicated to IP-to-IP gateways.
“With this book, the authors provide an in-depth look at the breadth of voice gateway features and capabilities, as well as providing voice gateway configuration guidance.”
—Christina Hattingh, Access Technology Group, Cisco Systems®
Denise Donohue, CCIE® No. 9566, is a design engineer with AT&T. She is responsible for designing and implementing data and VoIP networks for SBC and AT&T customers.
David L. Mallory, CCIE No. 1933, is a technical education consultant with Cisco Systems, supporting Cisco voice certifications. Prior to this role, David was a systems engineer supporting several global enterprise customers.
Ken Salhoff, CCIE No. 4915, is a systems engineer with Cisco Systems. Ken has been specializing in voice technologies at Cisco for the past six years.
- Understand the pros and cons of MGCP, H.323, and SIP, how they implement call signaling and describe call flow, and how to configure each protocol
- Learn the various analog and digital voice circuit options used to connect a VoIP network to the PSTN
- Configure and troubleshoot PSTN, PBX, and IP WAN connections
- Build scalable dial plans and understand the different types of dial peers
- Understand the various ways gateways control called and calling phone numbers
- Examine call admission control (CAC) techniques
- Configure Class of Restrictions (CoR) for both inbound and outbound calls
- Deploy and troubleshoot SRST and MGCP fallback
- Evaluate DSP considerations and resources
- Support interactive voice response (IVR) and advanced call-handing applications using Tcl scripts and VXML
- Deploy and configure basic and advanced gatekeeper functions
- Configure and troubleshoot IP-to-IP gateways
This IP communications book is part of the Cisco Press® Networking Technology Series. IP communications titles from Cisco Press help networking professionals understand voice and IP telephony technologies, plan and design converged networks, and implement network solutions for increased productivity.
Table of Contents
Foreword
Introduction
Part I Voice Gateways and Gatekeepers
Chapter 1 Gateways and Gatekeepers
The Role of Voice Gateways
Types of Voice Gateways
The Role of Voice Gatekeepers
The Role of IP-to-IP Gateways
Introduction to Voice Protocols
Media Gateway Control Protocol
H.323
Session Initiation Protocol
Skinny Client Control Protocol
Real-Time Transport Protocol
Call Control Agents
Cisco CallManager
Cisco CallManager Express
SIP Proxy Server
Cisco Enterprise Gateway
PBX with Toll Bypass
Deployment Scenarios
Single Site Deployment
Multisite with Centralized Call Control
Multisite Deployment with Distributed Call Control
Case Study: Introduction
Chapter Review Questions
Part II Gateways
Chapter 2 Media Gateway Control Protocol
Introduction to MGCP
Pros
Cons
MGCP Operation
MGCP Messages
Registering with CallManager
Call Flow with MGCP
Call Flow Between Analog Phones
ISDN Connections with Backhaul
MGCP Fallback
Dial Plan Considerations
Implementing MGCP Gateways
Basic MGCP Gateway Configuration
Configuring MGCP Fallback
Assigning an MGCP Source IP Address
Configuring MGCP PRI and BRI Backhaul
Enabling Multicast Music on Hold
Configuring Cisco CallManager
Configuring CallManager Redundancy
Configuring DTMF Relay
Securing MGCP Gateways
Troubleshooting Tools
Case Study: Configuring an MGCP Gateway
Review Questions
Chapter 3 H.323
H.323 Specifications
H.323 Network Components
H.323 Gateways
H.323 Gatekeepers
H.323 Terminals
Multipoint Control Units
H.323 Proxy Servers
Call Flow
H.323 Fast Start
H.323 Protocol Pros and Cons
Pros
Cons
When to Use H.323
Dial Plan Considerations
Implementing H.323 Gateways
Voice Class Configuration
Voice Service VoIP Configuration
Toll Bypass
Defining H.323 Gateways on CallManager
Redundancy
DTMF Relay
Securing H.323 Gateways
Troubleshooting Tools
Case Study: Configuring an H.323 Gateway
Review Questions
Chapter 4 Session Initiation Protocol
Description of SIP
SIP Functional Components
SIP Messages
SIP Call Flow
Call Flow Between Two SIP Gateways
Call Flow Using a Proxy Server
Call Flow Using Multiple Servers
Call Flow Using Cisco CallManager 5.x
SIP Pros and Cons
Pros
Cons
When to Use SIP
Dial Plan Considerations
Implementing SIP Gateways
SIP Dial Peer Configuration
SIP UA Configuration
SIP Voice Service Configuration
Toll Bypass
Registering with CallManager
DTMF Relay
Securing SIP Gateways
Allowing H.323 to SIP Connections
Troubleshooting Tools
Case Study: Configuring SIP Between a Gateway and CallManager 5.x
Review Questions
Chapter 5 Circuit Options
Circuit Signaling
Analog Circuits
FXS/FXO
E&M
Digital Circuits
T1
E1
E1 R2
ISDN
Echo Cancellation
Review Questions
Chapter 6 Connecting to the PSTN
PSTN Circuit Selection Overview
Supported Analog Connection Types
Supported Digital Connection Types
Analog Trunks
Configuring FXO Connections
Configuring DID Connections
Configuring Centralized Automated Message Accounting Connections
Caveats and Restrictions
Digital Trunks
Configuring E1/T1 Physical Layer Connections
Configuring ISDN PRI Trunks
Configuring E1 R2 Trunks
Configuring T1 CAS Trunks
Configuring ISDN BRI Trunks
Case Study: Add an E1 R2 Connection to the Leeds Gateway
Review Questions
Chapter 7 Connecting to PBXs
Analog Trunks
Configuring FXO/FXS Connections
Configuring E&M Trunks
Digital Trunks
Configuring E1/T1 Physical Layer Connections
Configuring ISDN PRI Trunks
Configuring E1 R2 or T1 CAS Trunks
Configuring Transparent Common Channel Signaling
Case Study: Implementing a Cisco Voice Gateway at the Shanghai Office
Review Questions
Chapter 8 Connecting to an IP WAN
Applications for Connecting to an IP WAN
Design Considerations
Quality of Service
Using Class Maps to Classify Traffic
Using Policy Maps
Mapping to MPLS Classes
Link Fragmentation and Interleave
Compression
AutoQos
Providing Fax and Modem Services
Providing Fax Services
Providing Modem Services
Security
Securing Voice Media and Signaling
V3PN
NAT and VoIP
Firewalls and VoIP
Case Study: Using a T1 Link as a Tie Line
Review Questions
Chapter 9 Dial Plans
Numbering Plans
Private Numbering Plans
PSTN Numbering Plans
Overlapping Numbering Plans
Building a Scalable Dial Plan
Dial Peers
Inbound Versus Outbound Dial Peers
Dial Peer Matching
Inbound Dial Peer Matching
Outbound Dial Peer Matching
Verifying Dial Peers
Outbound Dial Peer Targets
POTS Versus VoIP Outbound Dial Peers
Dial Peer Operational Status
Dial Peers Versus Cisco CallManager
Case Study: Configuring PSTN Access
Review Questions
Chapter 10 Digit Manipulation
Basic Digit Manipulation
Digit Stripping
Forward Digits
Prefix Digits
Number Expansion
Voice Translation Rules and Profiles
Creating Voice Translation Rules
Building Regular Expressions
Creating Voice Translation Profiles
Applying Voice Translation Profiles
Blocking Calls Using Voice Translation Rules and Profiles
Testing Voice Translation Rules
Manipulating Caller ID
CLID Commands
Station ID Commands
Order of Operation in Digit Manipulation
Troubleshooting Digit Manipulation
Case Study
Review Questions
Chapter 11 Influencing Path Selection
Hunt Groups
Using the preference Command
Using the huntstop Command
Using Digit Manipulation
Using Trunk Groups
Tail-End Hop-Off
Call Admission Control
Local CAC Mechanisms
Measurement-Based CAC Mechanisms
Resource-Based CAC Mechanisms
Resource Reservation Protocol
POTS-to-POTS Call Routing Considerations
Case Study: Implementing Gateway-Controlled RSVP
Review Questions
Chapter 12 Configuring Class of Restrictions
COR Overview
COR Operation
Implementing COR
Assigning COR Lists with SRST
Assigning COR Lists with Cisco CallManager Express
Assigning COR Lists to SIP Phones with CallManager Express
Restricting Inbound Calls
Case Study: Implementing COR for Miami
Review Questions
Chapter 13 SRST and MGCP Gateway Fallback
SRST Overview
Fallback Time
Restoral Time
Configuring SRST
Gateway Configuration
CallManager Configuration
Dial Plan Considerations
Planning
Configuring SRST Dial Plan Patterns
SRST Features
Auto Attendant
Maximum Line Appearances
Conferencing
Transferring Calls
Forwarding Calls
Voice-Mail Integration
Music on Hold
SIP SRST
Configuring SIP Registrar Server
Configuring a Voice Register Pool
Call Preservation
Secure SRST
Configuring Secure SRST
MGCP Gateway Fallback
Configuring MGCP Gateway Fallback
Verifying and Troubleshooting SRST
Verifying and Troubleshooting MGCP Gateway Fallback
Case Study: Integrating SRST with an Analog Voice-Mail System
Review Questions
Chapter 14 DSP Resources
Need for DSP Resources
Determining the DSP Resources Required
DSP Types
Voice Termination
DSP Sharing
Transcoding and MTP Resources
Conference Bridge Resources
Configuring DSP Resources
Configuring Transcoding and Conferencing (C549)
Configuring Enhanced Transcoding and Conferencing (C5510)
Transcoding for CallManager Express
Case Study: Add DSP Resources to the Miami Gateway
Review Questions
Chapter 15 Using Tcl Scripts and VoiceXML
Tcl IVR and VoiceXML Application Overview
Programming Resources
Sample Applications
Auto Attendant
Basic ACD
Fax Detect
T.37 Store and Forward Fax
Malicious Call ID
Cisco Voice Portal
Embedded Event Manager
Downloading Tcl Scripts from Cisco.com
Configuring the Gateway to Use a Tcl Script
Initializing Tcl Scripts and Specifying Parameters
Applying TclScripts
Tcl Packages and Parameter Namespaces
Tcl Parameters in Cisco IOS Release 12.3(14)T and Above
Implementing the AA Tcl Script
Creating Audio Files
Restrictions and Caveats
Case Study: Implementing ACD Application
Review Questions
Part III Gatekeepers
Chapter 16 Deploying Gatekeepers
Gatekeeper Functionality
Gatekeeper Signaling
RAS Signaling
Gatekeeper Update Protocol
Gatekeeper Transaction Message Protocol
E.164 Number Resolution
Zone Prefixes
Technology Prefixes
Gatekeeper Call Routing Process
Call Admission Control
Gatekeeper Deployment Models
Redundancy
Resource Availability Indicator
Directory Gatekeeper
Gatekeepers with CallManager
Security with Gatekeepers
Tokenless Call Authentication
Review Questions
Chapter 17 Gatekeeper Configuration
Configuring Basic Gatekeeper Functionality
Configuring Gatekeeper Zones
Configuring Gateways to Use H.323 Gatekeepers
Technology Prefixes
Configuring Zone Prefixes and Dial Peers
Dynamic Prefix Registration
Configuring Call Admission Control
Multiple Gatekeeper Configurations
Configuring Directory Gatekeepers
Troubleshooting Gatekeepers
Registration Issues
Call Routing Issues
CallManager and Gatekeepers
Configuring a CallManager Gatekeeper Trunk
Gatekeeper Redundancy
Hot Standby Routing Protocol
Gatekeeper Clustering
Load Balancing
Troubleshooting Gatekeeper Clustering
Configuring Resource Availability Indicator
Configuring Gatekeeper Security
Troubleshooting Gatekeeper Security
Case Study: Deploying Gatekeepers to Assist in Migration to VoIP
Review Questions
Part IV IP-to-IP Gateways
Chapter 18 Cisco Multiservice IP-to-IP Gateway
IP-to-IP Gateway Overview
Cisco Multiservice IP-to-IP Gateway
Architecture
Media-Handling Modes
Protocol Support
Basic Configuration
Via-Zones
IP-to-IP Gateway Features
Video Support
Address Hiding
Security
DTMF Interworking
Fax Support
Quality of Service
Call Admission Control
Transcoding
VXML and Tcl Scripts
Billing
show Commands
debug Commands
Case Study: Providing Enterprise VoIP Trunking to VoIP Service of the Service Provider
CallManager Configuration
Review Questions
Appendix A Answers to Chapter-Ending Review Questions
商品描述(中文翻譯)
描述
《Cisco Voice Gateways and Gatekeepers》是一本關於Cisco語音閘道和閘道器的書籍。該書詳細介紹了在實施VoIP網絡時遇到的真實問題的解決方案。這本實用指南幫助讀者了解Cisco閘道和閘道器的選擇和配置。書中深入探討了閘道的選擇、設計問題、功能配置以及安全性和高可用性問題。豐富的示例、屏幕截圖、配置片段和案例研究使這本書成為對於任何對於在VoIP網絡中正確實施閘道和閘道器感興趣的人來說都是一本真正實用和有用的指南。重點放在實際部署中遇到的最佳實踐和常見問題上。
《Cisco Voice Gateways and Gatekeepers》分為四個部分。第一部分概述了IP語音網絡。第二部分專門介紹了語音閘道,包括Media Gateway Control Protocol(MGCP)、H.323、Session Initiation Protocol(SIP)、語音電路選項、與PSTN、PBX和IP WAN的連接、撥號計劃、數字操作、路由選擇、限制類別、可恢復遠程站點電話(SRST)和MGCP回退、數字信號處理器(DSP)資源以及Tool Command Language(Tcl)腳本和Voice XML(VXML)。第三部分介紹了語音閘道器,包括詳細的部署和配置。第四部分專門介紹了IP到IP閘道。
“通過這本書,作者們深入探討了語音閘道的功能和能力,並提供了語音閘道的配置指南。”
- Christina Hattingh, Access Technology Group, Cisco Systems
作者介紹:
- Denise Donohue, CCIE No. 9566,是AT&T的設計工程師,負責為SBC和AT&T的客戶設計和實施數據和VoIP網絡。
- David L. Mallory, CCIE No. 1933,是Cisco Systems的技術教育顧問,支持Cisco語音認證。在擔任這個角色之前,David是支持幾家全球企業客戶的系統工程師。
- Ken Salhoff, CCIE No. 4915,是Cisco Systems的系統工程師。