Cisco press cisco voice gateways and gatekeepers aug 2006 ISBN 158705258x

Understand the pros and cons of MGCP, H.323, and SIP, how they implement call signaling and describe call flow, and how to configure eachprotocol Learn the various analog and digital voi

By Denise Donohue, David Mallory, Ken Salhoff

Publisher: Cisco Press Pub Date: August 17, 2006 Print ISBN-10: 1-58705-258-X Print ISBN-13: 978-1-58705-258-3 Pages: 648

as the interface between an IP telephony network and the PSTN or PBX

Gatekeepers and IP-to-IP gateways help these networks scale Gatekeepersprovide call admission control, call routing, address resolution, and bandwidthmanagement between H.323 endpoints including Cisco IOS® voice gatewaysand Cisco® Unified CallManager clusters IP-to-IP gateways allow VoIP calls

world problems encountered when implementing a VoIP network This practicalguide helps you understand Cisco gateways and gatekeepers and configure themproperly Gateway selection, design issues, feature configuration, and securityand high-availability issues are all covered in depth The abundant examples,screen shots, configuration snips, and case studies make this a truly practicaland useful guide for anyone interested in the proper implementation of

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 voicegateway 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 isresponsible for designing and implementing data and VoIP networks for SBC

David L Mallory, CCIE No 1933, is a technical education consultant withCisco Systems, supporting Cisco voice certifications Prior to this role, Davidwas a systems engineer supporting several global enterprise customers

Ken Salhoff, CCIE No 4915, is a systems engineer with Cisco Systems Kenhas 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 eachprotocol

Learn the various analog and digital voice circuit options used to connect aVoIP 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 phonenumbers

Examine call admission control (CAC) techniques

Configure Class of Restrictions (CoR) for both inbound and outbound callsDeploy 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

This IP communications book is part of the Cisco Press® Networking

Technology Series IP communications titles from Cisco Press help networkingprofessionals understand voice and IP telephony technologies, plan and designconverged networks, and implement network solutions for increased

By Denise Donohue, David Mallory, Ken Salhoff

Publisher: Cisco Press Pub Date: August 17, 2006 Print ISBN-10: 1-58705-258-X Print ISBN-13: 978-1-58705-258-3 Pages: 648

fitness is implied

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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

Trademark Acknowledgments

All terms mentioned in this book that are known to be trademarks or servicemarks have been appropriately capitalized Cisco Press or Cisco Systems, Inc.cannot attest to the accuracy of this information Use of a term in this bookshould not be regarded as affecting the validity of any trademark or servicemark

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At Cisco Press, our goal is to create in-depth technical books of the highestquality and value Each book is crafted with care and precision, undergoingrigorous development that involves the unique expertise of members from theprofessional technical community

Readers' feedback is a natural continuation of this process If you have anycomments regarding how we could improve the quality of this book, or

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Executive Editor Brett Bartow

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Copyright © 2003 Cisco Systems, Inc All rights reserved CCIP, CCSP, theCisco Arrow logo, the Cisco Powered Network mark, the Cisco Systems

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Dedications

Denise Donohue: This book is dedicated to my incredibly patient family,

especially my husband, Kevin, who has done a good, if not enthusiastic, job ofcooking dinners the past few months I'd also like to dedicate this book to all thepeople who have been so generous in sharing their knowledge with me over theyears

David Mallory: This book is dedicated to my wife, Tammy, and my children,Melissa and Laura, who sacrifice much for my endeavors

Ken Salhoff: This book is dedicated to my wife, Jan, and my children, Jeff andDani, who have been encouraging and supportive through this entire process.During the many hours I've spent locked in the lab, Jan has handled the addedfamily workload without complaint, and for that I will be forever grateful

Denise Donohue, CCIE No 9566, is a design engineer with AT&T She is

responsible for designing and implementing data and VoIP networks for SBCand AT&T customers Prior to that, she was a Cisco instructor and course

director for Global Knowledge Her CCIE is in Routing and Switching

David L Mallory, CCIE No 1933, is a technical education consultant with

Cisco Systems, Inc supporting Cisco voice certifications Prior to this role,

David was a systems engineer supporting several global enterprise customers.David has presented on voice gateways and gatekeepers at Networkers and hasachieved four CCIE certifications: Routing and Switching, WAN Switching,Security, and Voice

Ken Salhoff, CCIE No 4915, is a systems engineer with Cisco Systems, Inc.Ken has been specializing in voice technologies with Cisco for the past six years

In the systems engineering role, Ken has supported several global enterprisecustomers using Cisco voice technologies Ken has achieved two CCIE

certifications: Routing and Switching, and Voice

Contributing Author

Jayesh Chokshi has worked in the field of networking for about 10 years and hasbeen with Cisco Systems, Inc for the last 6 years He has worked in the field ofVoice over IP as part of the Solutions Engineering and Technical MarketingTeam and helped design and deploy multiple customer voice networks

Currently, he works as a Technical Marketing Engineer and focuses on CiscoMultiservice IP-to-IP Gateway product development and customer deployments

Danelle Au is a product manager in the Access Technology Group at Cisco

Systems She defines the strategy and roadmap and drives product requirements

in the areas of IP communications She was involved in the product managementand marketing activities for the introduction of the Cisco CallManager Express(CME) and Survivable Remote Site Telephony products She is currently

responsible for IP Communications security, the voice gateway module portfolio,and DSP technologies on the Cisco Integrated Services Routers Danelle is a co-author of the Cisco IP Communications Express book (ISBN 158705180X) andholds an M.S in Electrical Engineering from University of California, Berkeley

Christina Hattingh is a member of the technical staff in the branch office IPCommunications group at Cisco Systems The products in this group, includingthe Cisco 2600, 2800, 3600, 3700, and 3800 series platforms, were some of thefirst Cisco platforms to converge voice and data by offering PSTN and PBXvoice gateway interfaces and critical QoS features on WAN interfaces, whilemore recently integrating numerous IP network voice services, call control, andapplication elements such as conferencing/transcoding services, RSVP Agent,IP-IP gateway, Cisco CME, and Cisco UE into the router-based platform In thisrole, she helps guide development projects, trains Cisco sales staff and Ciscoresale partners on new router-based voice technologies, and advises customers

on voice network deployment and design

Sibrina Shafique, CCIE No 10871, is a member of the technical marketing teamthat works on the Cisco voice gateways She works closely with engineeringteams to guide voice-related projects and sales staff to provide training and

technical assistance In her previous role as a technical support engineer her taskwas to analyze, configure, and troubleshoot complex product and network

problems on customer networks on all Cisco Voice gateways and their

interaction with PSTN, third-party PBX systems She has end-to-end

understanding of voice networks and has assisted with large voice deployments

Lingling Zhang has been working in Cisco Systems as Product Manager for fiveyears The primary focus of her role is to lead definition and development ofVoice over IP products and services in the portfolio of Cisco access routers anddevelop Cisco competitive strategy and positioning in the low-end access

market Before joining Cisco, Lingling Zhang worked in China Network World

Brandon Ta, CCIE No 2494, is a Consulting Systems Engineer for the

Enterprise East Voice Group at Cisco Systems, Inc Brandon worked for CiscoSystems, Inc for nine years and has been focusing on unified communication forthe last six years He has worked on some of larger IP Telephony designs andnetworks that span from 16,000 to more than 100,000 Cisco IP phones Besidethe CISSP certification and a handful of other technical certifications, Brandonhas four CCIEs and was the first in the world to receive three CCIEs back in

2000, where he was featured in Cisco Packet magazine Brandon has presented

at the Cisco IP Telecommunication User Group (CIPTUG) conference as well asthe Cisco Networkers Conference on Unified Communication Technology

This book is the product of a few authors and many behind-the-scenes workerswho deserve recognition Our thanks go to our editor, Brett Bartow, who did hisbest to keep us on track and on time, in spite of ourselves Thanks also to

Christopher Cleveland and Dan Young, our development editors, for ensuringthat our words made sense The editing and production staff had the unenviablejob of making sure all spelling, grammar, punctuation, and illustrations werecorrect, and we are grateful for their work

Our technical editors on this book did a great job of making sure the book wastechnically correct and suggesting ways to improve the content The book wouldnot have been as good without them A big thank you to Danelle Au, ChristinaHattingh, Sibrina Shafique, Brandon Ta, and Lingling Zhang!

Denise would like to thank the AES Engineering and Callisma groups withinAT&T for the loan of lab equipment Thanks also to Tom Petzold for helpingkeep this relevant

David would like to thank the voice TEC team for their reviews and suggestionsand for their insights on instruction and certification He also wants to thank hismanager, Drew Rosen, for supporting him in this and all his other efforts

Ken would like to thank his manager, Michael Aaron, for his support during thisprocess

[View full size image]

The conventions used to present command syntax in this book are the sameconventions used in the Cisco IOS Command Reference The Command

Reference describes these conventions as follows:

Boldface indicates commands and keywords that you enter literally asshown In actual configuration examples and output (not general commandsyntax), boldface indicates commands that you input manually (such as ashow command)

Cisco IOS routers have shipped with voice interface cards since 1997, and afterthis capability was available the term voice gateway became part of the VoIPvernacular, seemingly overnight The voice interfaces allowed routers to provide

a critical interconnectivity link between the traditional data IP networks and thetraditional voice (PSTN, PBXs, and key systems) networks With this

technology, the industry widely built toll bypass networks during the late 1990s:Enterprises connected their PBXs at different sites with VoIP "trunks" instead ofwith TIE lines or the PSTN, and service providers leveraged IP backbone

networks to offer calling-card services and cut-rate long-distance and

international calling

VoIP did not fascinate the popular imagination of the likes of Jeff Pulver ofVoice-on-the Net (VON) and other industry observers until end-uservisibledevices, such as IP phones, and IP-based applications brought the technology tothe forefront Voice gateway technology is still the pale sibling of the IP

telephony world that creates no buzz, and yet it is also the workhorse of everysingle VoIP network Even as VoIP endpoints become ever more prevalent inbusinesses and residences, voice gateways still provide critical interconnectivitywith billions of traditional PSTN and PBX voice endpoints, without which

companies cannot operate their communications networks

Although the idea of a voice gateway is conceptually simple enoughit's a

demarcation between two networks and translates the protocols from one (theTDM world) to the other (the IP world)the technology has become increasinglysophisticated and the features more intertwined over the years Choosing the

"right" voice gateway and configuring the "right" set of features for a particularnetwork is no longer the task for the uninitiated The question I hear most

frequently is whether to deploy MGCP, H.323, or SIP gateways Cisco gatewaysare protocol agnostic and support all of these protocols and several variationsthereof, and the answer to the question posed is not a simple one: The optimalnetwork design depends on a large number of considerations Some protocolsand designs are better suited to particular types of networks, partly owing to thearchitecture of the protocols themselves and partly due to the features that havebeen chosen for implementation over the years

A Cisco Press book with comprehensive coverage focused entirely on voice

features and capabilities, as well as providing voice gateway configuration

guidance The book explains the major VoIP protocols, MGCP, H.323, and SIP,their structure and operation, and the considerations to choose among them Itdiscusses in detail the PSTN and PBX circuit connection technologies and

choices There are often multiple connection choices on the central office orPBX switch as well as on the voice gateway side of the circuit, and which ofthese would provide the features, cost points, and manageability that are optimalfor your network might not be obvious at first glance

The book goes on to provide insights into many other areas of gateway selectionand deployment, including the myriad choices in carrying fax and modem trafficover IP, dial plan features and digit manipulation tools, call admission controlcapabilities to keep voice traffic off the IP network when it does not have thequality levels to carry it, a review of DSP technology and operation, and anexamination of IP connectivity implications and QoS features required to carryvoice traffic with decent quality Later chapters in the book also include

discussions on pure IP-oriented topics such as TCL and VXML applicationscapabilities, conference mixing, transcoding, gatekeeper functions and

connectivity, and IP-to-IP gateway (session border controller) services and

features All of these pure VoIP services are offered by the same platforms thatare also voice gateways, even though TDM connectivity is not a necessary

component for the IP-only services

The book also covers key areas of interest in any network, including securitymeasures and high availability VoIP network security is a wide topic fully

deserving of its own book-length treatment, but this book provides enough basicinformation to get your network deployed It covers how voice gateway trafficpasses through firewalls and NAT devices, how to encrypt voice signaling andmedia traffic to or from a voice gateway, as well as configuring class of servicerestrictions such that certain call patterns are allowed while others are blockedper the policy of your network High availability is essential in all networksachapter in this book is dedicated to the discussion of how gateways fail overwhen other network components are out of contact, as well as how gatewayfeatures interoperate with IP Phone failover features such as SRST to maintaindial tone and PSTN network access for your end users at all times

Throughout the book is a case study that solidifies the chapter discussions by

providing practical, hands-on examples of how the configuration of the systemimplements the features This, together with the detailed chapter-by-chaptercoverage of crucial gateway topics, make this an invaluable book essential to thetool chest of anyone contemplating the implementation of a new network,

The use of Voice over IP (VoIP) is rapidly expanding You can find all sorts ofglowing statistics about the adoption of VoIP by businesses and consumers.Traditional telecommunications companies are phasing in IP voice in parts oftheir backbone Service providers are lowering the cost to businesses of

implementing VoIP by offering hosted and managed solutions Organizationsboth large and small are embracing VoIP

Voice gateways are an essential part of VoIP networks Gatekeepers and IP-to-IPgateways help these networks scale Yet no definitive reference is available onthese subjects A network engineer who needs to design a VoIP network or

configure these devices has to search to find the necessary information Thisbook strives to fill that need as much as humanly possible It builds on the

foundation laid by the Cisco CVoice course, focusing on advanced deploymentoptions and features of Cisco IOS-based voice gateways and gatekeepers

In a field that is constantly changing and developing, great care has been taken

to make this book relevant and accurate The authors hope that you will find it agood learning tool and a good resource for implementing and troubleshootingyour voice gateways and gatekeepers

Goals and Methods

The goal of this book is to produce a practical guide to help people both

understand Cisco gateways and gatekeepers and configure them properly Thisbook seeks to describe the Cisco voice gateway and gatekeeper theories andprotocols in a way that readers can apply to their networks It provides examples,screen shots, configuration snips, and case studies to make this a useful guideand a reference book The book includes the typical tasks and issues in

deploying voice gateways and gatekeepers, in addition to advanced features andcapabilities Emphasis is placed on the accepted best practices and commonissues encountered It is not written specifically as a preparation for the CiscoGateway Gatekeeper exam, but the topics that are covered on the exam are

included in this book

Who Should Read This Book?

configuration, and maintenance of VoIP and IP Telephony networks Secondaryaudiences include network managers and certification candidates who would like

to augment their Cisco Gateway Gatekeeper exam preparatory material withmore in-depth information This book also serves as a resource to CCVP andCCIE voice candidates who are preparing for the written or lab exams

It is assumed that the reader understands IP networking and is familiar with thetopics covered in the following Cisco courses: CVoice, CIPT, QoS, and IPTT.This includes a basic understanding of VoIP and time-division multiplexing(TDM) voice fundamentals, the concepts and configuration of basic IP voicerouters, and Cisco CallManager basics

This book is designed to be flexible and allow you to easily move between

chapters to focus on topics of interest to you However, if you do intend to readevery chapter, the order in the book is an excellent sequence to use because itprogresses from basic to more advanced subjects Chapter 1 provides an

as routers, standalone devices, and switch modules, H.323 gatekeepers, and IP-Control Protocol (SCCP), and Real-Time Transport Protocol (RTP) protocols Italso describes different types of call agents that are used in IP voice networks Inaddition, it introduces the network used throughout the book in examples andcase studies

Part II, "Gateways"

Chapter 2, "Media Gateway Control Protocol" This chapter covers how MGCPimplements call signaling, describes call flow using MGCP, and discusses thepros and cons of the protocol and when to use it It then covers implementingMGCP gatewaysconfiguration of the router, addition of an MGCP gateway toCallManager, configuration of dual tone multifrequency (DTMF) relay, somesecurity features, and troubleshooting of MGCP gateways

Chapter 3, "H.323" This chapter discusses the implemention of call signalingwith the H.323 protocol, some of the protocols that are part of the H.323 suite,call flow using H.323, and the pros and cons of the protocol and when to use it

It then covers implementing H.323 gatewaysconfiguring the router, using tollbypass, adding a gateway to CallManager, configuring DTMF relay, allowingH.323-to-H.323 connections, configuring both H.323 and MGCP on a gateway,using H.323 security features, and troubleshooting

implements call signaling, describes the various SIP functions and how theyparticipate in the call flow, and lists the pros and cons of the protocol and when

to use it It covers implementing SIP gatewaysconfiguring the router, using SIPtoll bypass, adding the gateway to CallManager, using SIP security features,allowing SIP-to-H.323 connections, and troubleshooting SIP gateways

Chapter 5, "Circuit Options" This chapter examines the various analog and

digital circuits that gateways use to connect a VoIP network to the public

switched telephone network (PSTN) It looks at the different signaling types,features, and uses of each type of circuit; issues such as echo cancellation; andselection and sizing of the circuit

Chapter 6, "Connecting to the PSTN" PSTN connections are examined in moredepth in this chapter The analog coverage includes a discussion on configuringand troubleshooting Foreign Exchange Office (FXO), direct inward dial (DID),and Centralized Automated Message Accounting (CAMA) The digital coverageincludes explanations on configuring and troubleshooting T1 and E1 PRI, BRI,and channel-associated signaling (CAS) connections The chapter also discussescaller ID and name delivery considerations

Chapter 7, "Connecting to PBXs" This chapter covers the configuration of

analog PBX trunks using Ear and Mouth (E&M) and FXO interfaces, and digitalPBX trunks using CAS, PRI, and BRI interfaces It discusses Transparent

Common Channel Signaling (T-CCS) and Q Signaling (QSIG)

Chapter 8, "Connecting to an IP WAN" This chapter covers reasons for

connecting a VoIP network to an IP WAN, QoS considerations, use of the

modular quality of service command-line interface (MQC) to provide the neededquality of service, and QoS over an MPLS network Handling faxes and modems

in a VoIP network are discussed This chapter also describes the need for

securing voice traffic over a WAN by using SRTP and V3PN, the implications ofNAT on voice traffic, and the use of firewalls with VoIP

Chapter 9, "Dial Plans" This chapter covers dial plan design and implementation

It discusses building a scalable dial plan, handling overlapping number ranges,and implementing a dial plan on gateways and call agents It helps you

understand the different types of dial peers and how they affect call routing byexamining the way that the gateway selects dial peer matches

Chapter 11, "Influencing Path Selection" This chapter covers several differentways of influencing route selection, including hunt groups, trunk groups, and tailend hop-off The chapter examines call admission control (CAC) techniqueslocalCAC using maximum connections and local voice busyout; measurement-basedCAC using IP SLA, PSTN fallback, and advanced local voice busyout; and

resource-based CAC using Resource Reservation Protocol (RSVP) and RSVPagent, and using CAC with gatekeepers It also discusses considerations whenusing POTS-to-POTS call routing

Chapter 12, "Configuring Class of Restrictions" This chapter explains what COR

is and how it operates It covers using COR with CallManager Express (CME)implementations, and on a gateway with Survivable Remote Site Telephony(SRST), restricting both inbound and outbound calls

Chapter 13, "SRST and MGCP Gateway Fallback" This chapter gives an

overview of SRST with ways to adjust the failover and fallback times, and

shows how to configure SRST on both the gateway and CallManager It

describes MGCP fallback and how to configure it SRST dial plan considerationsare discussed, along with configuring SRST call features, configuring SIP SRST,preserving calls, and troubleshooting SRST and MGCP fallback

Chapter 14, "DSP Resources" This chapter examines codec considerations andconfiguration, digital signal processor (DSP) considerations, determination ofDSP resources needed, the DSP modes, and configuration of conferencing andtranscoding resources on both the gateway and CallManager

Chapter 15, "Using TCL Scripts and VoiceXML" This chapter explains how tosupport interactive voice response (IVR) and advanced call-handling

applications using Toolkit Command Language (Tcl) and VoiceXML It givesexamples, tells how to download Tcl scripts, and shows how to configure

gateways to use them This chapter also examines the auto attendant (AA) scriptand creation of audio files in more detail and discusses some caveats and

Part III, "Gatekeepers"

Chapter 16, "Deploying Gatekeepers" This chapter provides the conceptualinformation that you need to integrate gatekeepers into your VoIP network Itdescribes gatekeeper functionality in networks with and without a CallManager,gatekeeper CAC and address resolution, redundancy and load balancing,

gatekeeper security, and use of hierarchical gatekeepers A discussion of

planning a gatekeeper implementation includes placement, bandwidth, and dialplan issues Scalabilty and multizone enhancements are also addressed

Chapter 17, "Gatekeeper Configuration" This chapter is a practical guide togatekeeper configuration It covers router configuration for basic and advancedgatekeeper functions such as directory gatekeepers, gatekeeper redundancy,Resource Availability Indication (RAI), and security CallManager configurationfor gatekeeper use and gatekeeper redundancy is also discussed, along with ways

configuring a via-zone gatekeeper, and troubleshooting the configuration

Part I: Voice Gateways and Gatekeepers

Chapter 1 Gateways and Gatekeepers

This book will help you understand the roles of IP voice gateways and

gatekeepers and how to integrate them into your voice network

Gateways can be configured with many different features and functions and canbecome quite complex It is important for network engineers to understandgateway functions, be able to choose when to use which, and be able to

configure their gateways Engineers need a thorough understanding of

gatekeeper function and configuration to ensure proper call flow In addition,they must address issues of security and redundancy

The basic function of a gateway is to translate between different types of

networks In the data environment, a gateway might translate between a FrameRelay network and an Ethernet network, for example In a VoIP environment,voice gateways are the interface between a VoIP network and the public

switched telephone network (PSTN), a private branch exchange (PBX), or

analog devices such as fax machines In its simplest form, a voice gateway has

an IP interface and a legacy telephone interface, and it handles the many tasksinvolved in translating between transmission formats and protocols At least onegateway is an essential part of any IP telephony network that interacts with thePSTN or with analog devices In addition, when gateways are properly

configured, many can take over for a Cisco CallManager when it is unreachable

Figure 1-1 shows a simple VoIP network with a voice gateway connected to boththe VoIP network and the PSTN

Figure 1-1 Voice Gateway Example

The gateway allows communication between the two networks by performingtasks such as these:

Interfacing with the IP network and the PSTN or PBX

multiplexing (TDM) call control protocols

Performing call setup and teardown for calls between the VoIP and PSTNnetworks by terminating and reoriginating the call media and signaling.Providing supplementary services, such as call hold and transfer

Types of Voice Gateways

This book deals primarily with using Cisco routers as voice gateways, but anassortment of different types and models of Cisco voice gateways is available

Exchange Station (FXS) ports, and Ear and Mouth (E&M) ports Digital optionsinclude T1/E1 PRI ports, ISDN BRI ports, and T1 channel associated signaling(CAS) ports You can divide Cisco gateways into several categories, each withits own capabilities For more information on the current models, visit

www.Cisco.com

Routers

You can use many Cisco routers, such as 1700, 2600, 2800, 3600, 3700, 3800,

7200, and 7500 models, as voice gateways Voice gateway routers can containanalog ports such as FXO and FXS, and digital ports such as E1 and T1 Mostcan use the MGCP, SCCP, SIP, and H.323 protocols The remainder of this bookfocuses mainly on router-based voice gateways

Standalone Voice Gateways

Unlike routers, standalone voice gateways are used only as voice gateways.They come in two types: digital and analog

Digital trunk gateways, such as the AS5000 series interface, connect an IP

telephony network and the PSTN or a PBX via their trunk ports

Analog trunk gateways, such as AT-2 or AT-4, connect to the PSTN or a PBX viaanalog trunks Analog station gateways, such as an ATA186, VG224, or VG248,connect to analog devices such as telephones, fax machines, or voice-mail

systems The signaling protocols that are available vary by gateway model TheATA186, an analog-to-IP adapter, can support two analog devices each with itsown telephone number It has one Ethernet port that connects to the VoIP

network, and two voice ports for connecting analog devices It can be controlled

by Cisco CallManager and CallManager Express, and it supports SIP and SCCPprotocols However, CallManager 5.0 does not have native SIP support for theATA186

VG224 and VG248 are Cisco voice gateways with 24 and 48 analog FXS ports,respectively These are line-side gatewaysthey do not interface with the PSTN,but with end-user analog devices Both models allow a CallManager,

CallManager Express (CME), or an SRST router multiple to control analog

The analog telephones, faxes, or modems that are attached to the gateway appear

as individual endpoints to the CallManager You must configure the gateway onthe CallManager, and you must configure each port on CallManager with a

phone for each port in use

directory number and any call features it needs In CME, you must set up an e-The VG248 gateway uses only SCCP The VG224 gateway can use SCCP,

H.323, MGCP, and SIP CME interaction is supported only with the VG224gateway, not VG248 The VG224 gateway is Cisco-IOS based, whereas

Gatekeepers help VoIP networks scale to large sizes Companies that have

geographically dispersed voice networks, or networks that have become so largethat they are unwieldy, might opt to segment their network In a CallManagernetwork, you can create multiple clusters In that case, you would need to

configure a full mesh of connections over the IP WAN to link all the segments orclusters You would need to configure dial information for every remote location

on every gateway and CallManager cluster A better alternative is to use

gatekeepers In a network that has gatekeepers, trunks are needed only to thegatekeeper, and the gatekeeper maintains remote endpoint information

When you use gatekeepers, gateways and CallManagers register with their

gatekeeper Gatekeepers divide the network into "zones," or groups of devicesthat register with a particular gatekeeper When an H.323 gateway receives a callthat is destined to a remote phone, it queries the gatekeeper for the location ofthe endpoint If the call is destined for a different zone, you can configure thegatekeeper to allow it only if sufficient bandwidth is available In more complexnetworks, you can use a Directory gatekeeper to maintain information about allthe zones You can configure Cisco routers with the appropriate Cisco IOS asH.323 gatekeepers

Figure 1-2 shows an example of a company that has three CallManager clustersand a gatekeeper Each cluster has an intercluster trunk over the IP WAN to thegatekeeper Each cluster is its own zone

Figure 1-2 H.323 Gatekeeper Example

provides the following services:

Address resolution A gatekeeper resolves E.164 telephone numbers andH.323 IDs to endpoint IP addresses

Call admission control A gatekeeper permits or denies a call between

clusters

Bandwidth control A gatekeeper can refuse to admit calls that exceed theallocated bandwidth

Zone management A gatekeeper can register and manage endpoints withinits zone

Optional Security A gatekeeper can authenticate and authorize calls on anendpoint-by-endpoint basis

Optional call management A gatekeeper can maintain information about theendpoint call state