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9781587052583

Cisco Voice Gateways And Gatekeepers

by ; ;
  • ISBN13:

    9781587052583

  • ISBN10:

    158705258X

  • Edition: 1st
  • Format: Hardcover
  • Copyright: 2006-08-17
  • Publisher: Cisco Press
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Summary

Cisco Voice Gateways and Gatekeepers Understanding and configuring GW/GK in complex VoIP networks Denise Donohue, CCIEreg; 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 IOSreg; voice gateways and Ciscoreg; 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 Systemsreg; Denise Donohue, CCIEreg; 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. bull; bull;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 bull;Learn the various analog and digital voice circuit options used to connect a VoIP network to the PSTN bull; bull;Configure and troubleshoot PSTN, PBX, and IP WAN connections bull;Build scalable dial plans and understand the different types of dial peers bull;Understand the various ways gateways control called and calling phone numbers bull;Examine call admission control (CAC) techniques bull; bull;Configure Class of Restrictions (CoR) for both inbound and outbound calls bull;Deploy and troubleshoot SRST and MGCP fallba

Author Biography

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. 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 has achieved 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 enterprise customers using Cisco voice technologies. Ken has achieved two CCIE certifications: Routing and Switching, and Voice.

 

Table of Contents

Foreword xxiv
Introduction xxvi
Part I Voice Gateways and Gatekeepers
3(20)
Gateways and Gatekeepers
5(18)
The Role of Voice Gateways
5(3)
Types of Voice Gateways
7(1)
Routers
7(1)
Standalone Voice Gateways
7(1)
Switch Modules
8(1)
The Role of Voice Gatekeepers
8(2)
The Role of IP-to-IP Gateways
10(1)
Introduction to Voice Protocols
11(3)
Media Gateway Control Protocol
11(1)
H.323
12(1)
Session Initiation Protocol
12(1)
Skinny Client Control Protocol
13(1)
Real-Time Transport Protocol
14(1)
Call Control Agents
14(2)
Cisco CallManager
14(1)
Cisco CallManager Express
14(1)
SIP Proxy Server
15(1)
Cisco Enterprise Gateway
15(1)
PBX with Toll Bypass
15(1)
Deployment Scenarios
16(2)
Single Site Deployment
16(1)
Multisite with Centralized Call Control
17(1)
Multisite Deployment with Distributed Call Control
17(1)
Case Study: Introduction
18(3)
Chapter Review Questions
21(2)
Part II Gateways
23(434)
Media Gateway Control Protocol
25(28)
Introduction to MGCP
26(2)
Pros
27(1)
Cons
28(1)
MGCP Operation
28(5)
MGCP Messages
29(2)
Registering with CallManager
31(2)
Call Flow with MGCP
33(4)
Call Flow Between Analog Phones
33(2)
ISDN Connections with Backhaul
35(2)
MGCP Fallback
37(1)
Dial Plan Considerations
37(1)
Implementing MGCP Gateways
37(10)
Basic MGCP Gateway Configuration
38(1)
Configuring MGCP Fallback
39(1)
Assigning an MGCP Source IP Address
39(1)
Configuring MGCP PRI and BRI Backhaul
39(1)
Enabling Multicast Music on Hold
40(1)
Configuring Cisco CallManager
40(5)
Configuring CallManager Redundancy
45(1)
Configuring DTMF Relay
46(1)
Securing MGCP Gateways
47(1)
Troubleshooting Tools
47(2)
Case Study: Configuring an MGCP Gateway
49(2)
Review Questions
51(2)
H.323
53(26)
H.323 Specifications
53(1)
H.323 Network Components
54(2)
H.323 Gateways
55(1)
H.323 Gatekeepers
55(1)
H.323 Terminals
55(1)
Multipoint Control Units
56(1)
H.323 Proxy Servers
56(1)
Call Flow
56(2)
H.323 Fast Start
56(2)
H.323 Protocol Pros and Cons
58(1)
Pros
58(1)
Cons
59(1)
When to Use H.323
59(1)
Dial Plan Considerations
60(1)
Implementing H.323 Gateways
61(11)
Voice Class Configuration
62(2)
Voice Service VoIP Configuration
64(1)
Toll Bypass
65(2)
Defining H.323 Gateways on CallManager
67(2)
Redundancy
69(1)
DTMF Relay
70(2)
Securing H.323 Gateways
72(1)
Troubleshooting Tools
72(1)
Case Study: Configuring an H.323 Gateway
73(3)
Review Questions
76(3)
Session Initiation Protocol
79(36)
Description of SIP
79(9)
SIP Functional Components
81(1)
SIP Messages
82(6)
SIP Call Flow
88(7)
Call Flow Between Two SIP Gateways
88(2)
Call Flow Using a Proxy Server
90(2)
Call Flow Using Multiple Servers
92(1)
Call Flow Using Cisco CallManager 5.x
93(2)
SIP Pros and Cons
95(1)
Pros
95(1)
Cons
96(1)
When to Use SIP
96(1)
Dial Plan Considerations
96(1)
Implementing SIP Gateways
97(12)
SIP Dial Peer Configuration
98(1)
SIP UA Configuration
98(3)
SIP Voice Service Configuration
101(1)
Toll Bypass
102(1)
Registering with CallManager
102(1)
Configuring a SIP Trunk with CallManager 4.x
102(3)
Configuring a SIP Trunk with CallManager 5.x
105(2)
Configuring the Gateway for a CallManager Trunk
107(1)
DTMF Relay
107(2)
Securing SIP Gateways
109(1)
Allowing H.323 to SIP Connections
110(1)
Troubleshooting Tools
110(1)
Case Study: Configuring SIP Between a Gateway and CallManager 5.x
111(2)
Review Questions
113(2)
Circuit Options
115(30)
Circuit Signaling
115(1)
Analog Circuits
116(10)
FXS/FXO
116(1)
FXS
117(1)
FXO
117(1)
FXO Power Failover
117(1)
FXS/FXO Supervisory Signaling
117(1)
Address Signaling
118(1)
FXS-DID
119(1)
Informational Signaling
119(1)
Caller ID
120(1)
Supervisory Disconnect
121(1)
CAMA
122(1)
E&M
123(1)
Type I Signaling
124(1)
Type II Signaling
124(1)
Type III Signaling
124(1)
Type IV Signaling
125(1)
Type V Signaling
125(1)
Address Signaling
126(1)
Digital Circuits
126(16)
T1
127(2)
Channel-Associated Signaling
129(1)
E&M Signaling
129(1)
Feature Group D
130(1)
E1
131(1)
E1 R2
131(1)
Line Signaling
131(1)
Inter-Register Signaling
132(2)
ISDN
134(1)
ISDN Switch Configuration
134(2)
ISDN Call Signaling
136(2)
Information Elements
138(1)
Address Types
139(1)
NFAS
140(1)
QSIG
141(1)
Echo Cancellation
141(1)
Review Questions
142(3)
Connecting to the PSTN
145(38)
PSTN Circuit Selection Overview
145(2)
Supported Analog Connection Types
146(1)
Supported Digital Connection Types
147(1)
Analog Trunks
147(11)
Configuring FXO Connections
147(2)
Verifying and Troubleshooting
149(1)
Caveats and Restrictions
150(1)
Media Gateway Control Protocol
150(2)
Configuring DID Connections
152(2)
Verifying and Troubleshooting
154(1)
Caveats and Restrictions
154(1)
Configuring Centralized Automated Message Accounting Connections
154(2)
Verifying and Troubleshooting
156(1)
Caveats and Restrictions
157(1)
Digital Trunks
158(21)
Configuring E1/T1 Physical Layer Connections
158(1)
Verifying and Troubleshooting
159(3)
Configuring ISDN PRI Trunks
162(2)
MGCP
164(1)
Verifying and Troubleshooting
165(3)
Configuring E1 R2 Trunks
168(3)
MGCP
171(1)
Verifying and Troubleshooting
171(1)
Configuring T1 CAS Trunks
172(2)
MGCP
174(1)
Verifying and Troubleshooting
174(1)
Configuring ISDN BRI Trunks
175(2)
MGCP
177(1)
Verifying and Troubleshooting
178(1)
Case Study: Add an E1 R2 Connection to the Leeds Gateway
179(2)
Review Questions
181(2)
Connecting to PBXs
183(36)
Analog Trunks
184(6)
Configuring FXO/FXS Connections
184(2)
MGCP
186(1)
Configuring E&M Trunks
187(1)
Verifying and Troubleshooting
188(2)
Digital Trunks
190(17)
Configuring E1/T1 Physical Layer Connections
190(1)
Clocking
191(4)
Verifying and Troubleshooting
195(1)
Configuring ISDN PRI Trunks
196(2)
Verifying and Troubleshooting
198(3)
MGCP
201(2)
Verifying and Troubleshooting
203(1)
Q Signaling
204(2)
Configuring E1 R2 or T1 CAS Trunks
206(1)
Configuring Transparent Common Channel Signaling
207(6)
Case Study: Implementing a Cisco Voice Gateway at the Shanghai Office
213(4)
Review Questions
217(2)
Connecting to an IP WAN
219(56)
Applications for Connecting to an IP WAN
219(1)
Design Considerations
220(1)
Quality of Service
221(28)
Using Class Maps to Classify Traffic
222(1)
Classifying at Layer 4
222(3)
Classifying at Layer 3
225(1)
Classifying at Layer 2
226(1)
Using Policy Maps
227(5)
Mapping to MPLS Classes
232(7)
Link Fragmentation and Interleave
239(1)
Multilink PPP
240(2)
Frame Relay FRF.12 Fragmentation
242(2)
Frame Relay Fragmentation Using MLPoFR
244(1)
MLPoATM
245(1)
Compression
246(2)
AutoQos
248(1)
Providing Fax and Modem Services
249(11)
Providing Fax Services
250(1)
Configuring Cisco Fax Relay
251(1)
Configuring T.38 Fax Relay for MGCP Gateways
252(2)
Configuring T.38 Fax Relay for H.323 and SIP Gateways
254(1)
Configuring Super G3 Fax Relay
255(1)
Configuring Fax Passthrough
256(1)
Configuring Fax Passthrough for MGCP Gateways
256(1)
Configuring Fax Passthrough for H.323 and SIP Gateways
257(1)
Providing Modem Services
258(1)
Configuring Cisco Modem Relay
258(1)
Configuring Modem Passthrough
259(1)
Security
260(6)
Securing Voice Media and Signaling
261(1)
Securing Voice Media with Secure RTP
261(2)
Securing Voice Signaling with IPsec
263(1)
Securing Voice Signaling with TLS
263(1)
V3PN
264(1)
NAT and VoIP
265(1)
Firewalls and VoIP
265(1)
Case Study: Using a T1 Link as a Tie Line
266(6)
Review Questions
272(3)
Dial Plans
275(28)
Numbering Plans
276(5)
Private Numbering Plans
276(1)
Private Numbering Plan Design Considerations
276(2)
PSTN Numbering Plans
278(1)
North American Numbering Plan
279(1)
UK National Numbering Plan
279(2)
Overlapping Numbering Plans
281(2)
Building a Scalable Dial Plan
283(1)
Dial Peers
283(2)
Inbound Versus Outbound Dial Peers
284(1)
Call Legs
284(1)
Dial Peer Matching
285(13)
Inbound Dial Peer Matching
285(3)
Outbound Dial Peer Matching
288(1)
direct-inward-dial Is Not Configured
288(1)
direct-inward-dial Is Configured
289(1)
Verifying Dial Peers
289(2)
Outbound Dial Peer Targets
291(1)
Trunk Groups
291(2)
Dial Peer Hunting
293(1)
POTS Versus VoIP Outbound Dial Peers
293(1)
Dial Peer Operational Status
294(1)
Dial Peers Versus Cisco CallManager
295(3)
Case Study: Configuring PSTN Access
298(2)
Review Questions
300(3)
Digit Manipulation
303(24)
Basic Digit Manipulation
303(3)
Digit Stripping
304(1)
Forward Digits
305(1)
Prefix Digits
306(1)
Number Expansion
306(1)
Voice Translation Rules and Profiles
307(10)
Creating Voice Translation Rules
308(1)
Building Regular Expressions
309(1)
Using Regular Expressions When Rerouting a Call over the PSTN
310(2)
Using Regular Expressions to Delete Specific Digits
312(1)
Using Regular Expressions with Sets and Replacement Digits
312(1)
Using Regular Expressions to Change the Call Type or Numbering Plan
313(1)
Creating Voice Translation Profiles
314(1)
Applying Voice Translation Profiles
314(1)
Blocking Calls Using Voice Translation Rules and Profiles
315(1)
Testing Voice Translation Rules
316(1)
Manipulating Caller ID
317(1)
CLID Commands
317(1)
Station ID Commands
318(1)
Order of Operation in Digit Manipulation
318(2)
Troubleshooting Digit Manipulation
320(1)
Case Study
321(2)
Review Questions
323(4)
Influencing Path Selection
327(38)
Hunt Groups
328(4)
Using the preference Command
328(1)
Using the huntstop Command
329(1)
Using Digit Manipulation
330(2)
Using Trunk Groups
332(1)
Tail-End Hop-Off
333(1)
Call Admission Control
334(25)
Local CAC Mechanisms
336(1)
Physical DS0 Limitation
336(1)
Maximum Connections
336(1)
Local Voice Busyout
337(1)
Measurement-Based CAC Mechanisms
337(1)
IP SLA
338(1)
Calculated Planning Impairment Factor and Mean Opinion Scores
338(2)
Configuring IP SLA
340(2)
IP Service Assurance Agent and Response Time Reporter
342(2)
PSTN Fallback
344(2)
Advanced Voice Busyout
346(1)
Resource-Based CAC Mechanisms
346(1)
Local Gateway Resources
346(1)
Gatekeeper Zone Bandwidth
347(1)
Resource Availability Indication
348(1)
Resource Reservation Protocol
348(1)
Gateway-Controlled RSVP
349(2)
Configuring Gateway-Controlled RSVP
351(1)
CallManager-Controlled RSVP with RSVP Agent
352(2)
Configuring RSVP Agent
354(4)
RSVP and the IntServ/DiffServ Model
358(1)
Incorporating RSVP into the Voice Network
358(1)
POTS-to-POTS Call Routing Considerations
359(1)
Case Study: Implementing Gateway-Controlled RSVP
360(2)
Review Questions
362(3)
Configuring Class of Restrictions
365(16)
COR Overview
365(1)
COR Operation
366(1)
Implementing COR
367(3)
Assigning COR Lists with SRST
370(1)
Assigning COR Lists with Cisco CallManager Express
371(1)
Assigning COR Lists to SIP Phones with CallManager Express
372(1)
Restricting Inbound Calls
372(2)
Case Study: Implementing COR for Miami
374(5)
Review Questions
379(2)
SRST and MGCP Gateway Fallback
381(30)
SRST Overview
381(2)
Fallback Time
382(1)
Restoral Time
383(1)
Configuring SRST
383(4)
Gateway Configuration
383(3)
CallManager Configuration
386(1)
Dial Plan Considerations
387(3)
Planning
387(1)
H.323 Gateways
388(1)
MGCP Gateways
388(1)
Direct Extension Dialing
389(1)
Configuring SRST Dial Plan Patterns
389(1)
SRST Features
390(7)
Auto Attendant
390(1)
Maximum Line Appearances
391(1)
Conferencing
391(1)
Transferring Calls
391(1)
Transfer-Pattern
391(1)
Consultative Transfers
392(1)
Forwarding Calls
392(1)
Voice-Mail Integration
393(1)
Integrating with CUE
393(1)
Integrating with a Centralized Voice-Mail System
393(3)
Music on Hold
396(1)
SIP SRST
397(1)
Configuring SIP Registrar Server
397(1)
Configuring a Voice Register Pool
397(1)
Call Preservation
398(1)
Secure SRST
399(4)
Configuring Secure SRST
399(4)
MGCP Gateway Fallback
403(1)
Configuring MGCP Gateway Fallback
404(1)
Verifying and Troubleshooting SRST
404(1)
Verifying and Troubleshooting MGCP Gateway Fallback
405(1)
Case Study: Integrating SRST with an Analog Voice-Mail System
405(3)
Review Questions
408(3)
DSP Resources
411(28)
Need for DSP Resources
411(1)
Determining the DSP Resources Required
412(12)
DSP Types
413(1)
PVDM Versus PVDM2
414(1)
Voice Termination
414(4)
Codec Complexity
418(1)
Configuring Codec Complexity
419(1)
Flex Complexity
420(1)
DSP Sharing
421(1)
Enabling DSP Sharing
422(1)
Transcoding and MTP Resources
422(1)
Conference Bridge Resources
423(1)
Configuring DSP Resources
424(10)
Configuring Transcoding and Conferencing (C549)
425(4)
Configuring Enhanced Transcoding and Conferencing (C5510)
429(5)
Transcoding for CallManager Express
434(1)
Case Study: Add DSP Resources to the Miami Gateway
435(2)
Review Questions
437(2)
Using Tcl Scripts and VoiceXML
439(18)
Tcl IVR and VoiceXML Application Overview
439(3)
Programming Resources
441(1)
Sample Applications
442(2)
Auto Attendant
442(1)
Basic ACD
443(1)
Fax Detect
443(1)
T.37 Store and Forward Fax
443(1)
Malicious Call ID
443(1)
Cisco Voice Portal
444(1)
Embedded Event Manager
444(1)
Downloading Tcl Scripts from Cisco.com
444(1)
Configuring the Gateway to Use a Tcl Script
445(3)
Initializing Tcl Scripts and Specifying Parameters
445(1)
Applying TclScripts
446(1)
Configuring Inbound Applications
446(1)
Configuring Outbound Applications
446(1)
Tcl Packages and Parameter Namespaces
446(1)
Tcl Parameters in Cisco IOS Release 12.3(14)T and Above
447(1)
Order of Parameter Precedence
448(1)
Upgrading to Cisco IOS Release 12.3(14)T and Later
448(1)
Implementing the AA Tcl Script
448(2)
Creating Audio Files
450(1)
Restrictions and Caveats
451(1)
Case Study: Implementing ACD Application
451(3)
Review Questions
454(3)
Part III Gatekeepers
457(92)
Deploying Gatekeepers
459(26)
Gatekeeper Functionality
459(2)
Gatekeeper Signaling
461(8)
RAS Signaling
461(3)
Gatekeeper Discovery Process
464(2)
H.323 Call Flows Using Gatekeepers
466(2)
Gatekeeper Update Protocol
468(1)
Gatekeeper Transaction Message Protocol
469(1)
E.164 Number Resolution
469(3)
Zone Prefixes
470(1)
Technology Prefixes
471(1)
Gatekeeper Call Routing Process
472(1)
Call Admission Control
472(4)
Gatekeeper Deployment Models
476(5)
Redundancy
476(2)
Resource Availability Indicator
478(1)
Directory Gatekeeper
479(2)
Gatekeepers with CallManager
481(1)
Security with Gatekeepers
482(1)
Tokenless Call Authentication
483(1)
Review Questions
483(2)
Gatekeeper Configuration
485(64)
Configuring Basic Gatekeeper Functionality
485(13)
Configuring Gatekeeper Zones
486(1)
Configuring Gateways to Use H.323 Gatekeepers
487(2)
Technology Prefixes
489(1)
Configuring Zone Prefixes and Dial Peers
490(4)
Dynamic Prefix Registration
494(2)
Configuring Call Admission Control
496(2)
Multiple Gatekeeper Configurations
498(2)
Configuring Directory Gatekeepers
500(3)
Troubleshooting Gatekeepers
503(5)
Registration Issues
503(2)
Call Routing Issues
505(3)
CallManager and Gatekeepers
508(5)
Configuring a CallManager Gatekeeper Trunk
508(5)
Gatekeeper Redundancy
513(15)
Hot Standby Routing Protocol
513(3)
Gatekeeper Clustering
516(4)
Load Balancing
520(1)
Troubleshooting Gatekeeper Clustering
521(1)
Configuring Resource Availability Indicator
522(6)
Configuring Gatekeeper Security
528(8)
Troubleshooting Gatekeeper Security
530(6)
Case Study: Deploying Gatekeepers to Assist in Migration to VoIP
536(10)
Review Questions
546(3)
Part IV IP-to-IP Gateways
549(28)
Cisco Multiservice IP-to-IP Gateway
551(26)
IP-to-IP Gateway Overview
551(1)
Cisco Multiservice IP-to-IP Gateway
552(4)
Architecture
553(1)
Media-Handling Modes
554(1)
Protocol Support
555(1)
SIP Networks
555(1)
H.323 Networks
555(1)
Protocol Interworking
556(1)
Basic Configuration
556(4)
Via-Zones
558(2)
IP-to-IP Gateway Features
560(10)
Video Support
560(1)
Address Hiding
561(1)
Security
562(1)
H.323 Deployments
562(1)
SIP Deployments: Transport Layer Security
562(1)
Digest Authentication
562(1)
DTMF Interworking
563(1)
Fax Support
564(1)
Quality of Service
565(1)
Call Admission Control
565(1)
RSVP-Based CAC
565(1)
Max Connections-Based CAC
565(1)
IP Call Capacity-Based CAC
565(1)
Thresholds Based on CPU, Memory, and Total Calls
566(2)
Transcoding
568(1)
VXML and Tcl Scripts
568(1)
Billing
568(1)
show Commands
569(1)
debug Commands
569(1)
Case Study: Providing Enterprise VoIP Trunking to VoIP Service of the Service Provider
570(4)
CallManager Configuration
571(3)
Review Questions
574(3)
Appendix A Answers to Chapter-Ending Review Questions 577(22)
Index 599

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Excerpts

Preface Foreword Cisco IOS routers have shipped with voice interface cards since 1997, and after this capability was available the term voice gateway became part of the VoIP vernacular, seemingly overnight. The voice interfaces allowed routers to provide a critical interconnectivity link between the traditional data IP networks and the traditional 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 of with 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 of Voice-on-the Net (VON) and other industry observers until end-user-visible devices, such as IP phones, and IP-based applications brought the technology to the 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 every single VoIP network. Even as VoIP endpoints become ever more prevalent in businesses and residences, voice gateways still provide critical interconnectivity with 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 enough--itrs"s a demarcation between two networks and translates the protocols from one (the TDM world) to the other (the IP world)--the technology has become increasingly sophisticated and the features more intertwined over the years. Choosing the "right" voice gateway and configuring the "right" set of features for a particular network 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 gateways are protocol agnostic and support all of these protocols and several variations thereof, and the answer to the question posed is not a simple one: The optimal network design depends on a large number of considerations. Some protocols and designs are better suited to particular types of networks, partly owing to the architecture of the protocols themselves and partly due to the features that have been chosen for implementation over the years. A Cisco Press book with comprehensive coverage focused entirely on voice gateway technology and features was a long time coming, and at last 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. The book explains the major VoIP protocols, MGCP, H.323, and SIP, their structure and operation, and the considerations to choose among them. It discusses in detail the PSTN and PBX circuit connection technologies and choices. There are often multiple connection choices on the central office or PBX switch as well as on the voice gateway side of the circuit, and which of these would provide the features, cost points, and manageability that are optimal for your network might not be obvious at first glance. The book goes on to provide insights into many other areas of gateway selection and deployment, including the myriad choices in carrying fax and modem traffic over IP, dial plan features and digit manipulation tools, call admission control capabilities to keep voice traffic off the IP network when it does not have the quality levels to carry it, a review of DSP technology and operation, and an examination of IP connectivity implications and QoS features required to carry voice traffic with decent quality. Later chapters in the book also include discussions on pure IP-oriented topics such as TCL and VXML applications capabilities, confe

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