Cisco Voice Gateways and Gatekeepers (Hardcover)
暫譯: Cisco 語音閘道器與閘道管理器 (精裝版)

David Mallory, Ken Salhoff, Denise Donohue

  • 出版商: Cisco Press
  • 出版日期: 2006-08-27
  • 售價: $2,880
  • 貴賓價: 9.5$2,736
  • 語言: 英文
  • 頁數: 648
  • 裝訂: Hardcover
  • ISBN: 158705258X
  • ISBN-13: 9781587052583
  • 相關分類: Cisco
  • 已絕版

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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 語音閘道與閘道管理器

理解並配置複雜 VoIP 網路中的 GW/GK

Denise Donohue, CCIE® No. 9566

David Mallory, CCIE No. 1933

Ken Salhoff, CCIE No. 4915

語音傳輸網路 (VoIP) 的部署持續快速進行。語音閘道是 VoIP 網路的重要組成部分,負責處理在傳輸格式和協議之間轉換的多項任務,並作為 IP 電話網路與 PSTN 或 PBX 之間的介面。閘道管理器和 IP-to-IP 閘道幫助這些網路擴展。閘道管理器提供呼叫接入控制、呼叫路由、地址解析和帶寬管理,涵蓋 H.323 端點,包括 Cisco IOS® 語音閘道和 Cisco® Unified CallManager 集群。IP-to-IP 閘道允許 VoIP 通話穿越不同的 IP 網路。

《Cisco 語音閘道與閘道管理器》提供了在實施 VoIP 網路時遇到的現實問題的詳細解決方案。這本實用指南幫助您理解 Cisco 閘道和閘道管理器並正確配置它們。閘道選擇、設計問題、功能配置以及安全性和高可用性問題都深入探討。豐富的範例、螢幕截圖、配置片段和案例研究使這本書成為任何對於在 VoIP 網路中正確實施閘道和閘道管理器感興趣的人的實用和有用的指南。重點放在公認的最佳實踐和在現實部署中遇到的常見問題上。

《Cisco 語音閘道與閘道管理器》分為四個部分。第一部分提供 IP 語音網路的概述。第二部分專注於語音閘道,包括對媒體閘道控制協議 (MGCP)、H.323、會話啟動協議 (SIP)、語音電路選項、連接到 PSTN、PBX 和 IP WAN、撥號計劃、數字操作、路由選擇、限制類別、可存活遠端站點電話 (SRST) 和 MGCP 回退、數位信號處理器 (DSP) 資源,以及工具命令語言 (Tcl) 腳本和語音 XML (VXML) 的討論。第三部分涉及語音閘道管理器,包括詳細的部署和配置。第四部分專注於 IP-to-IP 閘道。

“透過這本書,作者提供了對語音閘道功能和能力的深入了解,並提供語音閘道配置指導。”
—Christina Hattingh, Access Technology Group, Cisco Systems®

Denise Donohue, CCIE® No. 9566,是 AT&T 的設計工程師,負責為 SBC 和 AT&T 客戶設計和實施數據及 VoIP 網路。

David L. Mallory, CCIE No. 1933,是 Cisco Systems 的技術教育顧問,支持 Cisco 語音認證。在此角色之前,David 是支持多個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,過去六年專注於 Cisco 的語音技術。

- 理解 MGCP、H.323 和 SIP 的優缺點,如何實施呼叫信號並描述呼叫流程,以及如何配置每個協議
- 學習用於將 VoIP 網路連接到 PSTN 的各種類比和數位語音電路選項

- 配置和故障排除 PSTN、PBX 和 IP WAN 連接
- 建立可擴展的撥號計劃並理解不同類型的撥號對等
- 理解閘道如何控制被叫和主叫電話號碼的各種方式
- 檢查呼叫接入控制 (CAC) 技術

- 配置進出呼叫的限制類別 (CoR)
- 部署和故障排除 SRST 和 MGCP 回退
- 評估 DSP 考量和資源
- 使用 Tcl 腳本和 VXML 支持互動式語音回應 (IVR) 和高級呼叫處理應用
- 部署和配置基本及高級閘道管理器功能
- 配置和故障排除 IP-to-IP 閘道

這本 IP 通訊書籍是 Cisco Press® 網路技術系列的一部分。Cisco Press 的 IP 通訊書籍幫助網路專業人員理解語音和 IP 電話技術,規劃和設計融合網路,並實施網路解決方案以提高生產力。

目錄

前言

介紹

第一部分 語音閘道與閘道管理器

第一章 閘道與閘道管理器
語音閘道的角色
語音閘道的類型
語音閘道管理器的角色
IP-to-IP 閘道的角色
語音協議介紹
媒體閘道控制協議
H.323
會話啟動協議
瘦客戶端控制協議
即時傳輸協議
呼叫控制代理
Cisco CallManager
Cisco CallManager Express
SIP 代理伺服器
Cisco 企業閘道
具通話繞過的 PBX
部署場景
單一站點部署
集中呼叫控制的多站點
分散呼叫控制的多站點部署
案例研究:介紹
章節回顧問題

第二部分 閘道

第二章 媒體閘道控制協議
介紹