Cisco Voice Over IP (CVOICE)

This Cisco Voice Over IP (CVOICE) class is a 5 day class that is presented by Cisco training partners to their end customers. Channel Partners nationwide hire proven AMS Subject Matter Expert Cisco Certified Systems Instructors (CCSI’s) to teach on-site or on-line classes.

Prerequisites

  • Working knowledge of fundamental terms and concepts of computer networking to include LANs, WANs, and IP switching and routing
  • Basic internetworking skills taught in Interconnecting Cisco Network Devices (ICND), or equivalent knowledge
  • Ability to configure and operate Cisco routers and switches and to enable VLANs and DHCP
  • Knowledge of traditional public switched telephone network (PSTN) operations and technologies

The audience for this course is:

  • Network administrators and network engineers
  • CCNP Voice candidates
  • Systems engineers

Upon completing this course, you will be able to;

  • Explain what a voice gateway is, how it works, and describe its usage, components, and features
  •  Describe the characteristics and configuration elements of VoIP call legs
  •  Describe how to implement IP phones using Cisco Unified Communications Manager Express
  •  Describe the components of a dial plan, and explain how to implement a dial plan on a Cisco Unified voice gateway
  •  Explain what gatekeepers and Cisco Unified Border Elements are, how they work, and what features they support
  •  Describe why QoS is needed, what functions it performs, and how it can be implemented in a Cisco Unified Communications network

Course Outline

Module 1: Introduction to VoIP

Lesson 1: Introducing VoIP

  • Describe the components of the Cisco Unified Communications architecture
  • Describe VoIP and the basic components of a VoIP network
  • Describe the major VoIP signaling protocols
  • Describe the differences between the gateway signaling protocols
  • Describe issues that can affect voice service in the IP network
  • Describe the characteristics of the protocols that are used for media transmission.

Lesson 2: Introducing Voice Gateways

  • Describe the functionality of gateways and their role of connecting VoIP to traditional PSTN and telephony equipment
  • Describe the different Cisco gateway platforms
  • Identify supported IP telephony deployment models
  • Identify the major characteristics and design guidelines of a single-site IP telephony deployment model
  • Identify the major characteristics and design guidelines of a multisite centralized IP telephony deployment model
  • Identify the major characteristics and design guidelines of a multisite distributed IP telephony deployment model
  • Identify the characteristics, limitations, and advantages of clustering over the IP WAN

Lesson 3: Specifying Requirements for VoIP Calls

  • Describe the factors that are present in IP networks that affect audio clarity
  • Describe MOS and PSQM and how they are used to measure audio quality
  • Describe QOS features as they relate to a VoIP network and the features of Cisco IOS software that deliver QoS throughout the network
  • Describe the challenge of transporting modulated data, including fax and modem calls, over IP networks
  • Describe how fax and modem pass-through, relay, and store and forward are implemented using Cisco IOS gateways
  • Describe how T.38 and pass-through are supported by H.323, SIP, and MGCP
  • Describe DTMF relay and how it is supported in MGCP, H.323 and SIP

Lesson 4: Understanding Codecs, Codec Complexity, and DSP Functionality

  • Describe various codecs and their bandwidth requirements
  • Describe how the number of voice samples that are encapsulated impacts bandwidth requirements
  • Calculate the overhead for Layer 2 and other protocols on a VoIP call
  • Use a formula to calculate the total bandwidth that is required for a VoIP call with and without VAD
  • Describe various types of DSPs, DSP functions and how DSPs are used as media resources
  • Describe codec complexity and where and how to configure it
  • Describe the DSP requirements for various media resources and show how calculate the actual number of required DSPs
  • Describe DSP farms, DSP farm profiles and how to configure conferencing and transcoding on a voice gateway
  • Describe the commands that are required to configure DSP farms on Cisco IOS gateways for enhanced media resources
  • Describe how to verify the correct operation of available media resources

Module 2: Voice Port Configuration

Lesson 1: Understanding Call Types

  • List the seven call types in a VoIP network
  • Describe the local call type
  • Describe the on-net call type
  • Describe the off-net call type
  • Describe the PLAR call type
  • Describe the PBX-to-PBX call type
  • Describe the intercluster trunk call type
  • Describe the on-net to off-net call type

Lesson 2: Configuring Analog Voice Ports

  • Describe the various types of voice port interfaces and where they are used
  • Describe the various types of analog interfaces and their characteristics
  • Describe how to configure three types of analog voice ports
  • Describe CAMA how to configure a voice port for CAMA
  • Describe how to configure voice ports for DID service
  • Describe timing configuration parameters on voice ports
  • Explain how to use show, test, and debug commands to verify analog voice port operation

Lesson 3: Understanding Dial Peers

  • Describe the functions of POTS and VoIP dial peers and call legs in relations to a simple VoIP network
  • Describe how gateways interpret call legs to establish end-to-end calls
  • Describe the functions of the POTS, VoIP, and default dial peers
  • Describe how to configure POTS dial peers
  • Describe how to configure VoIP dial peers
  • Explain how to use destination-pattern options to associate a telephone number with a given dial peer
  • Describe how the router matches inbound dial peers
  • Describe the default dial peer
  • Describe how the router matches outbound dial peers

Lesson 4: Configuring Digital Voice Ports

  • Describe the various types of digital voice ports
  • Describe T1 CAS trunks and associated signaling
  • Describe E1R2 CAS trunks and associated signaling
  • Describe ISDN
  • Describe ISDN signaling
  • Configure a T1 CAS trunk to the PSTN
  • Configure an E1 CAS trunk to the PSTN
  • Configure and verify BRI and PRI trunks to the PSTN
  • Verify digital voice port connections

Lesson 5: Understanding QSIG

  • Describe QSIG and its associated features
  • Describe how to configure QSIG support on Cisco IOS gateways
  • Describe how to verify QSIG trunks

Module 3: VoIP Gateway Implementation

Lesson 1: Implementing H.323 Gateways

  • Describe the functions that are performed by a typical H.323 gateway
  • Describe the advantages of H.323 as a voice gateway protocol
  • Describe the functional components that make up an H.323 environment
  • Describe the H.323 call establishment and maintenance process
  • Describe H.323 call signaling
  • Describe the types of multipoint conferences that are supported by H.323
  • Describe how to configure an H.323 gateway
  • Describe how to configure a single codec or codec negotiation on an H.323 gateway
  • Describe how to tune some H.323 timers
  • Configure fax pass-through and relay on H.323 gateways
  • Describe how to configure H.323 DTMF relay on an H.323 gateway
  • Describe how to verify the status of an H.323 gateway

Lesson 2: Implementing MGCP Gateways

  • Describe MGCP and its associated standards
  • Describe the advantages of MGCP as a voice gateway protocol
  • Describe the basic components of MGCP and their roles
  • Describe the basic concepts of MGCP
  • Describe the interactions between an MGCP call agent and its associated gateways
  • Configure an MGCP residential and trunk gateway on a Cisco router
  • Describe the commands that are used to verify an MGCP configuration

Lesson 3: Implementing SIP Gateways

  • Describe SIP and its related standards
  • Describe the advantages of SIP as a voice gateway protocol
  • Describe the functional and physical components of a SIP network
  • Describe three models for SIP call setup and disconnects: direct, using a proxy server, and using a redirect server
  • Describe the types, use, and structure of SIP messages
  • Describe SIP address formats, address registration, and address resolution
  • Describe special considerations for dealing with DTMF tones in a SIP environment
  • Configure SIP functionality on Cisco IOS gateways
  • Verify and troubleshoot a SIP gateway

Module 4: Dial Plan Implementation on Voice Gateways

Lesson 1: Understanding Dial Plans

  • Describe the characteristics and components of a dial plan
  • Describe the concept of endpoint addressing, including overlapping directory number issues
  • Describe the characteristics of call routing and the importance of path selection
  • Describe the characteristics of digit manipulation
  • Describe the characteristics of calling privileges
  • Describe the characteristics of call coverage
  • Describe the characteristics of a scalable dial plan in a VoIP network
  • Describe the requirements for PSTN dial plans in Cisco IOS environments and explain which dial plan components are important
  • Describe the special requirements for ISDN in Cisco IOS gateway deployments
  • Configure a PSTN dial plan on Cisco IOS gateways for inbound and outbound calls, including proper DNIS and ANI modification
  • Verify a PSTN dial plan on Cisco IOS gateways

Lesson 2: Implementing Numbering Plans

  • Describe the basic characteristics of a numbering plan
  • Describe the different types of numbering plans
  • Describe the attributes of a scalable numbering plan
  • Describe overlapping numbering plans and strategies to address the issue
  • Describe how to integrate internal and external PSTN numbering plans
  • Describe how to integrate existing dial plans into a VoIP network
  • Describe how VoIP operators can provide the location and telephone number of mobile callers to 911 operators
  • Implement a numbering plan

Lesson 3: Configuring Digit Manipulation

  • Describe basic digit manipulation and why you would need to use it
  • Describe digit stripping using a voice gateway
  • Describe digit forwarding on a voice gateway
  • Describe digit prefixing on a voice gateway
  • Describe number expansion on a voice gateway
  • Describe how a gateway collects and consumes digits and applies them to a dial peer
  • Describe CLID manipulation
  • Describe the capabilities of voice translation rules and profiles
  • Contrast voice translation profiles with the dialplan-pattern command
  • Create a dial peer with digit manipulation commands to divert calls that connect to a specified destination

Lesson 4: Configuring Path Selection

  • Describe how a call is routed and the correct path is selected
  • Describe how a router matches dial peers to determine path selection
  • Describe how the gateway matches information elements to dial peers
  • Describe some routing and path selection best practices
  • Describe various path selection strategies
  • Describe the characteristics of site-code dialing and toll bypass
  • Describe TEHO
  • Configure site-code dialing and toll bypass
  • Configure TEHO

Lesson 5: Implementing Calling Privileges on Cisco IOS Gateways

  • Describe calling privileges
  • Describe how COR can be used to implement calling privileges on Cisco IOS gateways
  • Describe how COR can be used in SRST and Cisco United Communications Manager Express environments
  • Configure COR on a Cisco IOS gateway using Cisco Unified Communications Manager Express and SRST

Module 5: H.323 Gatekeeper

Lesson 1: Introducing Gatekeepers

  • Describe the functionality of gatekeepers in an H.323 environment  
  • Define the hardware and software requirements for gatekeeper functionality
  • Describe the signaling between gateways and gatekeepers
  • Describe how directory gatekeepers enhance the scalability of a network
  • Describe how gatekeeper zone prefixes are used for call routing
  • Describe how gatekeeper technology prefixes are used for call routing
  • Describe how gatekeepers perform address resolution and call routing in different scenarios
  • Describe how GKTMP works
  • Describe some commands that are used to verify H.323 gatekeeper operation

Lesson 2: Configuring Basic Gatekeeper Functionality

  • List the steps necessary to configure a multizone gatekeeper for local and remote zone call routing
  • Configure local and remote zones on a gatekeeper
  • Configure zone prefixes on a gatekeeper
  • Configure technology prefixes on a gatekeeper
  • Configure gateways to register with a gatekeeper
  • Configure dial peers for gatekeepers
  • Verify that H.323 endpoints are registered properly and calls are correctly routed across a single gatekeeper

Lesson 3: Implementing Gatekeeper-Based CAC

  • Describe bandwidth operation in a gatekeeper zone
  • Describe zone bandwidth calculation in a gatekeeper network
  • Configure zone bandwidth on a gatekeeper
  • Verify zone bandwidth operation on gatekeepers
  • Describe how RAI performs resource-availability in gatekeeper networks
  • Configure RAI in a gatekeeper network
  • Verify RAI operation in gatekeeper networks

Module 6: ITSP Connectivity

Lesson 1: Understanding Special Requirements for External VoIP Connections

  • Describe the functionality of a Cisco UBE
  • Describe how Cisco UBEs can be utilized in enterprise VoIP environments
  • Describe how protocol interworking is performed on Cisco UBEs
  • Describe how Cisco UBEs handle media flows
  • Describe how Cisco UBEs perform codec filtering
  • Describe how Cisco UBEs can be used to perform RSVP-based CAC
  • Describe how Cisco UBEs can be integrated with gatekeeper networks
  • Describe call flows involving Cisco UBEs

Lesson 2: Implementing a Cisco UBE

  • Describe the command that are used to enable protocol interworking
  • Configure H.323-to-H.323 interworking on a Cisco UBE
  • Configure H.323-to-SIP interworking on a Cisco UBE
  • Describe the commands that are used to configure media flow-around, media flow-through, and transparent codec pass-through
  • Configure transparent codec pass-through and media flow-around on a Cisco UBE
  • Configure a Cisco UBE to register with a via-zone gatekeeper
  • Verify Cisco UBE and via-zone gatekeeper operation
  • The lesson includes these topics:
  • Protocol Interworking Command
  • Configuring H.323-to-H.323 Interworking
  • Configuring H.323-to-SIP Interworking
  • Media Flow and Transparent Codec Commands
  • Configuring Transparent Codec Pass-Through and Media Flow-Around
  • Configuring Cisco UBEs and Via-Zone Gatekeepers
  • Verifying Cisco UBEs and Via-Zone Gatekeepers

Hands-on Lab Exercises

  • Lab 2-1: Configuring Analog Voice Ports
  • Lab 2-2: Configuring POTS Dial Peers
  • Lab 2-3: Configuring VoIP Dial Peers
  • Lab 2-4: Configuring Digital Voice Ports
  • Lab 3-1: Implementing H.323 Gateways
  • Lab 3-2: Implementing SIP Gateways
  • Lab 4-1: Implementing Numbering Plans
  • Lab 4-2: Implementing PSTN Dial Plans on Cisco IOS Gateways
  • Lab 4-3: Configuring Path Selection
  • Lab 4-4: Implementing Calling Privileges on Cisco IOS Gateways
  • Lab 5-1: Configuring Basic Gatekeeper Functionality
  • Lab 5-2: Implementing Gatekeeper-Based CAC

To Hire a proven AMS Cisco Voice Subject Matter Expert and Cisco Certified Systems Instructor who teaches this class, Call 800-798-3901 Today!

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