CVoice (Cisco Voice over Frame Relay, ATM and IP)

Cisco Voice over IP (CVOICE), version 4.0 is a five-day instructor-led course that focuses on introducing students to the latest Cisco-based technologies for carrying voice transmissions over a variety of data networks. Upon completion of this course a student will be able to:

bulletDescribe the similarities and differences between traditional public switched telephone network (PSTN) voice networks and IP Telephony solutions
bulletExplain the processes and standards for voice digitization, compression, digital signaling, and Fax transport as they relate to Voice over IP (VoIP) networks
bulletConfigure voice interfaces on Cisco voice-enabled equipment for connection to traditional, non-packetized telephony equipment
bulletConfigure the call flows for Plain Old Telephone Service (POTS), VoIP, and default
dial peers
bulletDescribe the fundamentals of VoIP and identify challenges and solutions regarding its implementation
bulletCompare centralized and de-centralized call control and signaling protocols.
bulletDescribe specific Voice Quality issues and the Quality of Service (QoS) solutions used to
solve them
bulletApply fundamental knowledge of VoIP to dial plans and enterprise and service provider applications

This course is directly related to Cisco's AVVID solution.

To register call 916-852-2570
This course can be delivered by the methods below:
Classroom Learning $2795 USD
Self-Paced e-Learning See Below

You Learn...

bulletMotivation for voice and data integration
bulletVoice-capable Cisco routers
bulletApplications for voice capable Cisco routers
bulletAnalog and digital voice characteristics
bulletVoice encodings
bulletVoice signaling
bulletVoice Over IP environments (SIP, H.323, MGCP)
bulletQuality of Service techniques for "non-quality-of-service" networks (such as IP and Frame Relay)
bulletIntegrated voice/data network design and optimization
bulletVoice/data network configuration and troubleshooting
bulletLegacy voice technologies
bulletEmerging voice technologies

Who Would Benefit

This course will be of benefit to Network Engineers, Network Architects and Network Support professionals who

bulletInstall and configure voice and data network devices
bulletTake requests to add voice and data services/applications to the network
bulletPerform traffic analysis
bulletDesign networks to maintain a high quality of service in an integrated voice/data environment

 

Course Outline

Please choose below for specific Outline information.

Classroom Outline
Self Paced e-Learning Details

Classroom Outline

1. Introduction to Packet Voice Technologies

bulletTraditional telephony networks
bulletPacket voice networks
bulletIP data networks

2. Analog and Digital Voice Connections

bulletBasics of analog and digital voice
bulletProcesses and standards of voice digitization, compression, and digital signaling
bulletSignaling methods
bulletISDN voice interfaces
bulletSignaling between PBXs,
bulletCommon Channel Signaling (CCS) systems
bulletInternetworking of signaling systems
bulletFax and modem usage over a VoIP network

3. Configuring Voice Interfaces

bulletAnalog and digital voice interfaces
bulletAnalog and digital voice ports for optimal voice quality

4. Voice Dial Peers

bulletCall flows
bulletInbound and outbound dial peers
bulletApplication of voice dial peers
bulletSpecial purpose connections

5. Introduction to Voice over IP

bulletFundamentals of VoIP
bulletDifferences and similarities between VoIP and Voice over other technologies, such as Frame Relay or ATM
bulletRoles of Gateways in integrating VoIP with the traditional voice technologies found in enterprise and service provider networks
bulletVoIP protocol stack
bulletApplied headers
bulletUse of Real-Time Transport Protocol compressed (cRTP)
bulletBandwidth requirements for various codecs and data links
bulletMethods to reduce bandwidth consumption
bulletImplications of implementing security measures in VoIP networks

6. Voice over IP Signaling and Call Control

bulletTypes of various signaling
bulletCall control models
bulletCall control services
bulletFunctional components of H.323
bulletFunctional components of SIP
bulletFunctional components of MGCP
bulletH.323, SIP, and MGCP call control models

7. Improving and Maintaining Voice Quality

bulletVoice Quality Measurement including codec choice, which affects quality
bulletTransporting real-time voice in a non-real-time IP internetwork
bulletQuality of Service (QoS) functional areas and tools
bulletCampus networks
bulletWAN
bulletEffect of network design on QoS
bulletOvercoming jitter
bulletOvercoming delay
bulletCall Admission Control tools
bulletResource Reservation Protocol (RSVP)
bulletBusy-hour bandwidth allocation for both voice and data traffic

8. Scalable Numbering and Applications

bulletImplementing a scalable numbering plan
bulletCost-saving applications

Course Labs

Lab 1: Configure the Classroom Network

List the data interfaces and voice ports on your routers. Access the client/servers in all other pods in the classroom.

Lab 2:Voice Port Configuration

Customize and verify analog port operations and digital port operations.

Lab 3:POTS Dial Peers

Configure dial peers for locally terminated calls, PBX calls, and public switched telephone network (PSTN) calls.  Determine appropriate method of digit forwarding and manipulation.Create hunt groups and determine hunting behavior.

Lab 4:   Connections

Simulate auto attendant functions through the use of PLAR/PLAR OPX.

Use appropriate show and debug commands to monitor and troubleshoot the connections

Lab 5:Basic VoIP

Configure VoIP connections.  Verify basic call setup through debug commands.

Use appropriate show and debug commands to monitor and troubleshoot the connections.

Lab 6:VoIP with H.323

Configure single zone and multizone H.323 Gatekeeper environments for VoIP scalability.Use debug to show commands to monitor the status and progress of call setup procedures in an H.323 environment.

Lab 7: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 8:VoIP with MGCP

Configure your routers as MGCP Residential Gateways 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 9:Quality of Service

Implement quality improvements on low speed links with QoS features such as fragmentation, interleave, and Frame Relay traffic shaping. Implement features such as voice packet marking (tagging) and queueing to improve end-to-end voice quality.

Lab 10:Call Admission Control

Implement call admission control by setting the dial peer maximum connections, by way of RSVP and by using H.323 Gatekeeper.

Self-Paced e-Learning Details

Essentials of Cisco Voice (ECV)

2 Titles Product # 281167

Essentials of Cisco Voice (ECV) is a two-title course designed to teach you how to use the latest CiscoŽ technology to integrate voice over Frame Relay, ATM, and IP. You will determine the optimum service selection, equipment, and configuration of a voice-over-data network. Learn detailed analog and digital telephony fundamentals and principles of voice communications. Design branch and regional office voice connectivity using Cisco 3620/3810 multiservice equipment and configure the Cisco 2600 and 3600 for voice over IP.

*This course offers material comparable to the Cisco CVOICE class.

Title 1: Cisco Router Telephony Interfaces

Product # 281159
bulletVoice over Multiservice Networks
bulletVoice over IP capable Routers
bulletThe MC3810
bulletTraditional Corporate telephony Networks
bulletMultiservice Router Roles
bulletThe FXO and FXS Interface
bulletFXO and FXS Trunk Signaling
bulletE and M Trunk Signaling
bulletCodecs
bulletDigital Voice Lines
bulletIOS Configuration of Voice Interfaces
bulletIOS Configuration of Digital Voice Interfaces

Title 2: Voice Over IP, Frame Relay, and ATM

Product # 281160
bulletQuality of Service on Multiservice Networks
bulletVoice of Frame Relay Concepts
bulletVoice over Frame Relay Configuration
bulletThe Multi-Flex Trunk
bulletVoice Over ATM
bulletVoice Over IP
bulletVoIP Call Setup and Configuration
bulletThe H.323 Gatekeeper
bulletBuilding a Multiservice Network Case Study
bulletThe Cisco IP Telephony System
bulletCisco Phone Protocols
bulletCIPT H.323 Devices

 

Suggested Prerequisites

Interconnecting Cisco Network Devices (ICND) and Building Scalable CiscoŽ Internetworks (BSCI) are suggested prior to attending this course.

bulletICND (Interconnecting Cisco Network Devices)
bulletBSCI (Building Scalable Cisco Internetworks)

 

Suggested Follow-ons

Students followed up CVoice (Cisco Voice over Frame Relay, ATM and IP) by attending these popular classes:

bulletCIPT (Cisco IP Telephony)
bulletCUSAE (Cisco Unity System Administration/Engineering)

 

Self-Paced e-Learning

bulletTitle 01: Cisco Router Telephony Interfaces (281159) - $495 USD
bulletTitle 02: Voice Over IP, Frame Relay, and ATM (281160) - $495 USD
bulletEssentials of Cisco Voice (ECV) (281167) - $795 USD

 

Certifications

bulletCCIEŽ (CiscoŽ Certified Internetwork Expert) Communications & Services
bulletCiscoŽ IP Telephony Support Specialist
 

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916-852-2570

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