Implementing Multiple-Site Deployments

This chapter covers the following topics:

  • Describing why call admission control is important to maintain voice QoS across an IP WAN
  • Describing how the locations feature in Cisco CallManager provides call admission control for centralized call-processing environments
  • Configuring locations-based call admission control in a centralized call-processing deployment to limit the number of active calls and prevent oversubscribing the bandwidth on the IP WAN links
  • Explaining what AAR is, how it works, and the requirements for configuring Cisco CallManager for AAR
  • Describing how a gatekeeper can reduce the number of intercluster trunks that are required in a distributed call-processing environment
  • Identifying the communication procedures between an H.323 gatekeeper and H.323 endpoint, including discovery, registration, admission, and bandwidth requests
  • Configuring gatekeeper-based call admission control in a distributed call-processing deployment to limit the number of active calls and prevent oversubscribing the bandwidth on the IP WAN links
  • Describing how SRST provides Cisco CallManager failover capabilities
  • Configuring SRST on a supported gateway and in Cisco CallManager so that the SRST router assumes call-processing duties should the WAN link fail

Implementing multiple-site IP telephony deployments over an IP WAN requires additional planning to ensure the quality and availability of voice calls. When an IP WAN connects IP telephony clusters, a mechanism must exist to control the audio quality and video quality of calls over the IP WAN link by limiting the number of calls that are allowed on that link at the same time. Call admission control is the mechanism that ensures that voice calls do not oversubscribe the IP WAN bandwidth and affect voice quality.

When the priority queue of IP WAN bandwidth is consumed, a mechanism must exist to automatically reroute calls over the public switched telephone network (PSTN) without requiring the caller to hang up and redial the called party. Automated Alternate Routing (AAR) is a Cisco CallManager feature that automatically reroutes calls through the PSTN or other networks when priority bandwidth is insufficient in a centralized call-processing deployment.

Note

Referring to the priority queue of IP WAN bandwidth does not necessarily encompass the entire amount of WAN bandwidth available. Rather, it encompasses the amount of bandwidth you have set aside for high-priority traffic (including voice).

If connectivity with Cisco CallManager is lost, Cisco IP Phones become unusable for the duration of the failure. Cisco Survivable Remote Site Telephony (SRST) overcomes this problem and ensures that the Cisco IP Phones offer continuous service by providing call handling support for Cisco IP Phones directly from the Cisco SRST router.

This chapter describes the operation and configuration of call admission control, AAR, and SRST.

Part I: Cisco CallManager Fundamentals

Introduction to Cisco Unified Communications and Cisco Unified CallManager

Cisco Unified CallManager Clustering and Deployment Options

Cisco Unified CallManager Installation and Upgrades

Part II: IPT Devices and Users

Cisco IP Phones and Other User Devices

Configuring Cisco Unified CallManager to Support IP Phones

Cisco IP Telephony Users

Cisco Bulk Administration Tool

Part III: IPT Network Integration and Route Plan

Cisco Catalyst Switches

Configuring Cisco Gateways and Trunks

Cisco Unified CallManager Route Plan Basics

Cisco Unified CallManager Advanced Route Plans

Configuring Hunt Groups and Call Coverage

Implementing Telephony Call Restrictions and Control

Implementing Multiple-Site Deployments

Part IV: VoIP Features

Media Resources

Configuring User Features, Part 1

Configuring User Features, Part 2

Configuring Cisco Unified CallManager Attendant Console

Configuring Cisco IP Manager Assistant

Part V: IPT Security

Securing the Windows Operating System

Securing Cisco Unified CallManager Administration

Preventing Toll Fraud

Hardening the IP Phone

Understanding Cryptographic Fundamentals

Understanding the Public Key Infrastructure

Understanding Cisco IP Telephony Authentication and Encryption Fundamentals

Configuring Cisco IP Telephony Authentication and Encryption

Part VI: IP Video

Introducing IP Video Telephony

Configuring Cisco VT Advantage

Part VII: IPT Management

Introducing Database Tools and Cisco Unified CallManager Serviceability

Monitoring Performance

Configuring Alarms and Traces

Configuring CAR

Using Additional Management and Monitoring Tools

Part VIII: Appendix

Appendix A. Answers to Review Questions

Index



Authorized Self-Study Guide Cisco IP Telephony (CIPT)
Cisco IP Telephony (CIPT) (Authorized Self-Study) (2nd Edition)
ISBN: 158705261X
EAN: 2147483647
Year: 2004
Pages: 329

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