CIPT2V8: Cisco IP Telephony Part 2 v8

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Overview

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Schedule:
Daytime
Run time:
5 Days
Location type:
Open

vat information, completion, maximum participants, and starting dateplaces are unknown for this course.

Prerequisites:

Delegates are required to meet the following prerequisites:

* Working knowledge of fundamental terms & concepts of computer networking to include LANs, WANs, and IP switching and routing. CCNA Recommended
* Ability to configure and operate Cisco routers and switches and to enable VLANs and DHCP. BCMSN Recommended
* Fundamental knowledge of converged voice and data networks. CVOICE Recommended
* Ability to configure voice interfaces on Cisco voice-enabled equipment for connection to traditional, nonpacketized telephony equipment and to configure the call flows for POTS and VoIP dial peers. CIPT1 Recommended

Course Description:

Implementing Cisco Unified Communications IP Telephony Par…

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Description

Prerequisites:

Delegates are required to meet the following prerequisites:

* Working knowledge of fundamental terms & concepts of computer networking to include LANs, WANs, and IP switching and routing. CCNA Recommended
* Ability to configure and operate Cisco routers and switches and to enable VLANs and DHCP. BCMSN Recommended
* Fundamental knowledge of converged voice and data networks. CVOICE Recommended
* Ability to configure voice interfaces on Cisco voice-enabled equipment for connection to traditional, nonpacketized telephony equipment and to configure the call flows for POTS and VoIP dial peers. CIPT1 Recommended

Course Description:

Implementing Cisco Unified Communications IP Telephony Part 2 (CIPT2) v8 prepares you for installing and configuring, a Cisco Unified Communications Manager solution in a multisite environment. This course focuses on Cisco Unified CallManager Release 8, the call routing and signaling component for the Cisco Unified Communications solution. It also includes H.323 and Media Gateway Control Protocol (MGCP) gateway implementation, the use of a Cisco Unified Border Element, and configuration of Survivable Remote Site Telephony (SRST), different mobility features, and voice security.

This course includes lab activities in which you will apply a dial plan for a multisite environment, configure survivability for remote sites during WAN failure and implement solutions to reduce bandwidth requirements in the IP WAN. You will also enable call admission control (CAC) and automated alternate routing (AAR), a feature that allows rerouting of calls over the public switched telephone network (PSTN) in case of no available bandwidth. There are labs for implementing Cisco Unified Communications Manager Device Mobility, Cisco Unified Communications Manager Extension Mobility, Cisco Unified Mobility, and voice security.

This course includes the following modules:

Multisite Deployments

  • Identifying Issues in a Multisite Deployment
  • Identifying Solutions for a Multisite Deployment
  • Implementing Multisite Connections Hardening the IP Phone
  • Implementing a Dial Plan for Multisite Deployments

Centralized Call-Processing Redundancy

  • Examining Remote Site Redundancy Options
  • Implementing SRST and MGCP Fallback
  • Implementing Cisco Unified Communications Manager Express in SRST Mode

Bandwidth Management and Call Admission Control

  • Implementing Bandwidth Management
  • Implementing Call Admission Control

Features and Applications for Multisite Deployments

  • Implementing Call Applications on Cisco IOS Gateways
  • Implementing Device Mobility
  • Implementing Extension Mobility Implementing Cisco Unified Mobility

Multisite Deployments

  • Understanding Cryptographic Fundamentals and PKI
  • Understanding Native Cisco Unified Communications Manager Security Features and Cisco Unified Communications Manager PKI Implementing Security in Cisco Unified Communications Manager

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