Cisco CVOICE Training Class Voice Over IP CCVP
 Training Class Description
Cisco Course v5.0 and CVF v1.0 | Prepares you for Cisco Exam 642-432 CVOICE
We have enhanced this comprehensive Authorized Cisco course to introduce you to
the latest Cisco Voice over IP (VoIP) technologies and give you the skills and techniques to optimize your Cisco hardware for real performance and savings
We have added the required CVF v1.0 "Cisco Voice over IP Fundamentals" content to the standard 4-day CVOICE course. We realize that comprehensive hands-on training
is fundamental to your goals, and we offer a unique, real-world CVOICE lab environment that teaches you how to build and test a tsophisticated 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 CVOICE hands-on labs to
include a real dial plan and Class of Service for calling out to the PSTN, branch offices, and PBXs. You'll configure CVOICE Gateways using three different call control protocols while mastering the
connection details for traditional telephony. The traditional telephony fundamentals are taught on our exclusive,
additional first day. You'll experience real-world connections to PBXs, Key Telephone Systems, and the Public
Switched Telephone network. You will configure your router/gateway equipment to connect to our public dial plan network, using different call control protocols and procedures, demonstrating what you learn in class.
Audience
Technical professionals responsible for VoIP including voice/data integration
Upon Completion of this course you will learn
- Exclusive - Similarities and differences between traditional public switched telephone network (PSTN) voice networks and IP Telephony solutions
- Configure the call flows for Plain Old Telephone Service (POTS), VoIP, and default dial peers
- How to choose between centralized and decentralized call control and signaling protocols
- Analog and digital voice characteristics
- Processes and standards for voice digitization, compression, digital signaling, and Fax transport as they relate to VoIP networks
- Voice encodings and signaling
- Quality of Service (QoS) techniques for "non-quality-of-service" networks (such as IP and Frame Relay)
- Voice Over IP environments (SIP, H.323, MGCP)
- Integrated voice/data network design and optimization
- Voice/data network configuration and troubleshooting
- Emerging voice technologies
- Voice quality issues and the QoS solutions used to solve them
CCVP Certification Training Boot Camp
Cisco CVOICE Training Class Outline Voice over IP
.Voice Telephony Basics
- Home Subscriber Service
- Operation of the local loop
- Central Office responsibilities
- Supervisory signals
- Call Progress signals
- Address Signals
- Business Subscriber Service
- Four types of business systems
- Compare and contrast PBX and Key Systems
- Two types of Cisco Call Manager Systems
- Analog Trunking
- Digital Trunking
- Analog Interfaces
- Using FXS
- Using FXO
- Using E & M
- Digital Voice Coding
- Nyquist Theorem
- Standards for Waveform Coding
- Standards for Source Coding
- Compare coding rates and types
- Voice Quality Measurement
- Digital Voice Interfaces
- CCS
- CAS
- Compare and contrast T1 and E1
- Q.931
- Q.SIG
2. Introduction to VoIP
- VoIP Network Technologies
- VoIP Network Architectures
- Building Scalable Dial Plans
- Calculating Bandwidth Requirements
- Allocating Bandwidth for Voice and Data Traffic
- Considering Security Implications for VoIP Networks
3. Configuring Voice Networks
- Configuring Router Voice Ports
- Adjusting Voice Interface Settings
- Configuring Dial Peers
- Configuring Voice Port Network Connections
4. VoIP Signaling and Call Control
- Introducing Signaling and Call Control
- H.323
- Deploying and Configuring H.323
- Configuring SIP
- Configuring MGCP
- Comparing Call Control Models
5. Improving and Maintaining Voice Quality
- Designing for Optimal Voice Quality
- Implementing CAC (Call Admission Control)
Labs
- Lab 1: Enhanced - Topology and Deployment
Wire the CVOICE classroom network and image the laptop servers for each workgroup. A base IP configuration
is placed on the student router/gateway, while you become familiar with extension mobility test profiles. This lab
takes less than 90 minutes, and no additional wiring is required for any later labs.Enhanced content: Extension mobility test profiles
- Lab 2: Lab Familiarity
List the data interfaces and voice ports on your routers. Access the client/servers in all other pods in the
classroom. Verify and configure IP addressing on all interfaces. Set up routing.
- Lab 3: Voice Port Configuration
Configure and verify analog port operations, including FXS and FXO. Test operation of analog interfaces.
Configure digital port operations, including T1 CAS and ISDN PRI. Test operation of both digital ports.
- Lab 4: Enhanced - Dial Plan Lab 1: POTS Dial Peers
Configure dial peers for locally terminated, PBX, and PSTN calls. Determine appropriate method of digit
forwarding and manipulation. Configure explicit preference for dial peers, Caller ID, Direct Inward Dial (DID), and
Private Line Automatic Ringdown (PLAR). Enhanced content: Caller ID and Direct Inward Dial; Dial peers for all PSTN destinations
- Lab 5: Enhanced - Dial Plan Lab 2: Hunt Groups
Configure dial peers for emergency 911 and 9,911 for PRI ISDN connection (preferred) and FXO connection
(backup in case of failure). Enhanced content: 911 backup path to the PSTN
- Lab 6: Dial Plan Lab 3: VoIP Dial Peers
Configure VoIP connections. Configure Loopback interfaces for session targets. Verify basic call setup through
debug commands. Use appropriate show and debug commands to monitor and troubleshoot the connections.
Configure various explicit codecs and experiment with mis-matched codecs. Explore, calculate, and verify proper
bandwidth requirements for various types of voice calls. Configure and investigate codec negotiation.
- Lab 7: Dial Plan Lab 4: Tie Lines
Configure and test a tie-line toll bypass connection using VoIP. Dial between PBX connections across the
configured tie-line connection.
- Lab 8: Gateways and Gatekeepers
Configure single zone and multizone H.323 gatekeeper environments for VoIP scalability. Use debug and show
commands to monitor the status and progress of call setup procedures in an H.323 environment. Each workgroup configures both gateway and gatekeeper functions.
- Lab 9: VoIP with SIP
Use SIP direct (UA to UA) and proxy call control procedures to establish VoIP calls. Use debug commands to
analyze direct and proxy call control procedures.
- Lab 10: VoIP with MGCP
Configure your routers as MGCP Residential Gateways controlled by a Cisco CallManager and have the routers
use an MGCP call agent to establish voice between them. Use debug commands to analyze the interactions between MGCP Gateways and a call agent. Use show commands to view the status of MGCP endpoints,
connections, and calls.
- Lab 11: Quality of Service
Implement quality improvements on low speed links using AutoQoS. Experiment with QoS features such as
fragmentation, interleave, and Frame Relay traffic shaping. Implement features such as voice packet marking (tagging) and queuing to improve end-to-end voice quality.
- Lab 12: Call Admission Control
Implement call admission control three different ways: by setting the dial peer maximum connections, by way of
RSVP, and by using H.323 gatekeeper CAC
Class Dates and Locations
10/1/2007-10/5/2007 Toronto, ON
10/1/2007-10/5/2007 Calgary, AB
10/15/2007-10/19/2007 Vancouver, BC
11/5/2007-11/9/2007 Montreal, QC
11/19/2007-11/23/2007 Ottawa, ON
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