5-day Course on IP Telephony Technology
Cisco Course v6.0 | Cisco Unified Communications Manager Software v6.1 | Prepares you for Cisco Exam 642-456 CIPT2 v6.0.

Course Description
Prepare to install and configure a Cisco Unified Communications Manager (CUCM) solution in a multisite environment in this course. You'll focus on CUCM Release 6.0, the call routing and signaling component for the Cisco Unified Communications solution. Learn to implement H.323 and Media Gateway Control Protocol (MGCP), use a Cisco Unified Border Element, and configure Survivable Remote Site Telephony (SRST), mobility features, and voice security.
In lab activities, 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 enable call admission control (CAC) and automated alternate routing (AAR), a feature that allows rerouting of calls over the public switched telephone network (PSTN) if no bandwidth is available. You'll also work through labs for implementing CUCM Device Mobility, CUCM Extension Mobility, Cisco Unified Mobility, and voice security .
Why take CIPT2 from Us?
We offer a unique, real-world CIPT2 lab environment that teaches you how to build and test a sophisticated IP telephony network that you can use as a template for a real deployment. We have set ourselves apart from other Cisco training providers by enhancing our CIPT2 hands-on labs to include a real dial plan and Class of Service (CoS) for calling out to the PSTN, branch offices, and PBXs. No other training company offers a unique, real-world lab solution like ours.
Our voice network labs use the latest hardware and software.
- Train on Cisco's new Unified IP Phone 7965G
- Troubleshoot Cisco's new ISR routers: 2811s are used throughout the network
- Gain experience with recent stable IOS release: 12.3.14T3
- Configure CUCM 6.0
- Explore router DSP configuration: PVDM2-32 cards in every 2811 router
- Every router has 2xFXS, 2xFXO, and 2xT1 ports (PRI and T1-CAS) as well as serial ports for WAN connectivity
All of our IP telephony courses provide a simulated PSTN.
- Every pod has internal and external phones
- Build and test a real dial plan including:
- 911
- 3-digit service codes: 411, 511, etc.
- 7-digit local numbers: 681-1901
- 10-digit local numbers: 416-681-1901
- 11-digit long distance numbers: 1-733-802-1901
- International numbers: 011441902
And just like in a real network:
- The same simulated PSTN is accessible through both PRI and FXO ports
- The same simulated PSTN is accessible through all four clusters providing failover scenarios for bandwidth and connectivity problems
- Different area codes are deployed at all sites (two per cluster)
Build and test a real CoS solution that allows:
- Internal dialing
- 911 and service code dialing
- Local dialing
- Long distance dialing
- International dialing
Our labs provide comprehensive coverage of traditional telephony technologies.
- Configure FXS ports including Caller ID for calls to analog phones
- Configure FXO ports including Caller ID for calls from PSTN
- Configure connections to simulated PBXs via T1-CAS
Course Objectives
What you will learn:
- Issues in multisite deployments and their solutions
- Configure required dial plan elements
- Implement call-processing resiliency in remote sites using SRST, MGCP fallback, and CUCM Express
- Implement CAC to prevent oversubscription of the IP WAN
- Implement Cisco IOS Tcl and VoiceXML applications, along with mobility features such as CUCM Device Mobility, CUCM Extension Mobility, and Cisco Unified Mobility, so users are reachable via their office phone numbers, regardless of their physical location and the various devices they may use
- Secure a Cisco Unified Communications IP telephony deployment
Intended Audience
Network professionals who install, configure, and manage Cisco Unified Communications solutions. Cisco support personnel, channel partners, and customers benefit greatly from this course.
Prerequisites
Course Outline
- Course Introduction
- Multisite Deployments
- Issues in a Multisite Deployment
- Multisite Deployment Solutions
- Implementing Multisite Connections
- Implementing a Dial Plan for Multisite Deployment Solutions
- Centralized Call Processing Redundancy
- Examining Remote-Site Redundancy Options
- Implementing Cisco Unified SRST and MGCP Fallback
- Implementing CUCM Express in SRST Mode
- SRST using CME
- 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 CUCM Extension Mobility
- Implementing Cisco Unified Mobility
- IP Telephony Security
- Cryptographic Fundamentals and PKI
- Native CUCM Security Features and PKI
- Implementing Security in CU
Course Labs
- Lab 1: Topology and Deployment
- Lab 2: CUCM Initial Setup
- Lab 3: Gateway Lab 1: H.323 Trunks
- Lab 4: Gateway Lab 2: SIP Trunks
- Lab 5: Dial Plan Lab 1: Blocking Class of Service
- Lab 6: H.323 Configuration at the Branch
- Lab 7: Dial Plan Lab 2: H.323 Gateway - No CUCM
- Lab 8: Dial Plan Lab 3: H.323 Gateway - Routing via CUCM
- Lab 9: Tail-End Hop-Off
- Lab 10: SRST
- Lab 11: SRST using CME
- Lab 12: DSP Resources
- Lab 13: Regions and Device Pools
- Lab 14: Locations and AAR
- Lab 15: RSVP
- Lab 16: Gatekeepers
- Lab 17: Tool Command Language
- Lab 18: Device Mobility
- Lab 19: Extension Mobility
- Lab 20: Cisco Unified Mobility
- Lab 21: CUCM Authentication and Encryption
|