CIPTV1 - Implementing Cisco IP Telephony and Video Part 1 V1.0 (CIPTV1)
Cisco IP Telephony & Video, Part 1 (CIPTV1) v1.0 focuses primarily on Cisco Unified Communications Manager Version 10.x, which is the call-routing and signaling component for the Cisco Collaboration solution Lab exercises included in the course help learners to perform post-installation tasks, configure Cisco Unified Communications Manager, implement MGCP and H.323 and, SIP trunks, and build dial plans to place single site on-cluster and off-cluster calling for voice and video.
Level: COURSES.L_LEVEL_0
Total Hours： 8 小時
Number of Lessons： 5
開課日期： Sep 04, 2017
Day of the week： Thu
NTD 0

Course Introduction

Course Overview

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Who Should Attend

  • Network Professionals who install, configure and manage Cisco collaboration solutions.

Course Certifications

This course is part of the following Certifications:

Prerequisites

  • CICD - Implementing Cisco Collaboration Devices

  • CIVND2 - Implementing Cisco Video Network Devices V1.0

Course Objectives

  • How the CUCM administrative and service GUIs work

  • Activate, start, and stop CUCM services

  • Configure base CUCM components, such as date time groups, device pools, Call Manager groups, and other common elements

  • Add and delete phones manually and using auto registration

  • Add users, assign them capabilities, and associate them with phones

  • LDAP Integration including LDAP synchronization and LDAP authentication

  • LDAP attribute mapping and filters

  • Deploying IP Phone services

  • Configure phone features: Music on Hold (MOH) and phone services

  • Set up media resources to use for MOH and conferencing

  • Build a dial plan including route patterns, route lists, and route groups supporting both the NANP and variable-length dial plans

  • Deploy line/device Class of Service using partitions and calling search spaces for call blocking

  • Call hunting (hunt lists) and call queuing configuration

  • PSTN access methods, gateway vs. Cisco Unified Border Element (CUBE), and codec selection

  • PSTN access using MGCP gateways, including route lists, route groups, and digit manipulation

  • PSTN access using H.323 gateways including inbound and outbound dial peer selection

  • H.323 gateway digit manipulation, codec selection, and class of restriction

  • PSTN access using the CUBE and SIP trunks

  • CUBE and URI dialing

  • Media Resources including MOH, annunciators, and Media Termination Points (MTPs)

  • Hardware and software audio and video conference bridges

  • TelePresence MSE 800, TelePresence server, and TelePresence Conductor conferencing

  • Quality of Service (QoS) and bandwidth calculations

  • Best-Effort, IntServ, and DiffServ QoS models

  • QoS classification and marking

  • QoS policing and shaping

Course Content