CVOICE Implementing Cisco Voice Communications and QoS v8.0 Classroom

CVOICE_8 Implementing Cisco Voice Communications and QoS v8.0 

Course Description

Implementing Cisco Voice Communications and QoS (CVOICE) v8.0 teaches learners about voice gateways, characteristics of VoIP call legs, dial plans and their implementation, basic implementation of IP phones in Cisco Unified Communications Manager Express environment and essential information about gatekeepers and Cisco Unified Border Element. The course provides the learners with voice-related QoS mechanisms that are required in Cisco Unified Communications networks.

Prerequisites

To fully benefit from this course, it is recommended that you have the following prerequisite skills and knowledge:

• Working knowledge of fundamental terms and concepts of computer networking to include LANs, WANs, and IP switching and routing
• Ability to configure and operate Cisco IOS routers in IP environment at the CCNA level
• Basic knowledge of traditional voice, converged voice, and data networks at the CCNA Voice level

Associated Certifications

This course is a requirement for the CCNP Voice certification.

Who Should Attend

This course is intended for the following audience:

The primary audience for this course is as follows:

• Network Administrators and Network Engineers
• CCNP Voice candidates

The secondary audience for this course is as follows:

• Systems Engineers.

Course Objectives

After 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 Voice Gateways

Explain what a voice gateway is, how it works, and describe its usage, components, and features.

Lesson 1: Understanding Cisco Unified Communications Networks and the Role of Gateways

This lesson defines the characteristics and historical evolution of unified communications networks, the three operational modes of gateways, their functions, and the related call leg types. Upon completing this lesson, you will be able to meet these objectives:

• Describe the architecture and components of Cisco Unified Communications architecture
• Describe the function of voice gateways and their major roles in Cisco Unified Communications networks
• Identify the role of the gateways in four supported Cisco Unified Communications deployment models
• Briefly describe the different Cisco voice gateway platforms
• Identity the call legs that are created by a voice gateway in each operational mode

Lesson 2: Examining Gateway Call Routing and Call Legs

This lesson defines how gateways route calls and which configuration elements relate to incoming and outgoing call legs. Upon completing this lesson, the learner will be able to meet these objectives:

• Describe the functionality of gateways and their role of connecting VoIP to traditional PSTN and telephony equipment
• Describe the functions of POTS, VoIP dial peers, and call legs as components of a simple VoIP network
• Explain how gateways route calls end-to-end
• Describe how to configure POTS dial peers
• Explain how to use destination-pattern options to associate a telephone number with a given dial peer, and describe the number matching process
• Describe how the router matches inbound dial peers
• Describe how the router matches outbound dial peers
• Describe how the default dial peer is used in a gateway, when it is employed, and which default commands are used
• Explain the DID feature, describe the differences between two-stage and one-stage dialing, and explain what the DID feature does

Lesson 3: Configuring Gateway Voice Ports

This lesson defines how to connect a gateway to traditional voice circuits using analog and digital interfaces. Upon completing this lesson, you will be able to meet these objectives:

• Describe the factors that are present in IP networks that affect audio clarity
• Position the various types of analog and digital voice port interfaces in enterprise scenarios
• Describe the various types of analog voice ports and their characteristics
• Configure the analog voice ports
• List the types of digital voice ports and describe their major characteristics
• Describe ISDN and ISDN signaling
• Configure T1 and E1 trunks to the PSTN
• Configure ISDN PRI and BRI trunks\
• Tune various timers and parameters on analog and digital voice ports
• Explain and configure echo cancellation
• Verify analog and digital voice port configuration

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

This lesson defines DSPs and codecs, and explains different codec complexities and their usage. Upon completing this lesson, you will be able to meet these objectives:

• Explain voice codecs and their major features
• List voice quality evaluation methods and explain how they are applied to voice codecs
• Describe how the packet rate and protocol overhead impacts the total per-call bandwidth
• Explain various types of DSPs, DSP functions, and how DSPs are used in voice gateways
• Describe codec complexities
• Configure a DSP for voice termination at a voice gateway
• Monitor and verify operation of DSPs
• Lab 1-2: Configuring DSPs

Module 2: VoIP Call Legs

Describe the characteristics and configuration elements of VoIP call legs.

Lesson 1: Examining VoIP Call Leg Characteristics and VoIP Media Transmission

This lesson defines how VoIP signaling and media transmission differs from traditional voice circuits and explain how voice is sent over IP networks, including analog-to-digital conversion, encoding, packetization, and all variants of RTP. Upon completing this lesson, you will be able to meet these objectives:

• Describe how voice is transported over IP networks end-to-end, and compare traditional versus VoIP signaling and transmission mechanisms
• Describe the process of analog-to-digital voice conversion using PCM, including optional voice compression
• Describe the process of digital voice packetization and explain which options affect packet size
• Describe the characteristics of the four protocols used for media transmission in a VoIP network and outline their limitations
• Describe the process of suppressing silence to conserve per-call bandwidth, and list options that exist to accomplish this process

Lesson 2: Explaining H.323 Signaling Protocol

This lesson defines the characteristics of H.323 and explains when to use it. Upon completing this lesson, the learner will be able to meet these objectives:

• Describe the functional components of the H.323 environment, the functions that are performed by a typical H.323 gateway, and the advantages of H.323
• Describe the H.323 signaling stack, the signaling messages, and H.323 call flows on call setup and call teardown
• Describe the process of codec negotiation in H.323, the related protocols, and mechanisms such as Fast Start
• Describe how to configure an H.323 gateway
• Describe how to configure H.323 interface binding, transport protocol, and how to tune H.323 timers
• Describe major commands that are used to verify an H.323 gateway

Lesson 3: Explaining SIP Signaling Protocol

This lesson defines the characteristics of SIP, and explains when to use it. Upon completing this lesson, you will be able to meet these objectives:

• Describe SIP and its related standards, the functional and physical components of a SIP network, and the advantages of SIP as a voice gateway protocol
• Describe SIP signaling messages and the three models of SIP call setup: direct, using a proxy server, and using a redirect server
• Explain SIP address formats, address registration, and address resolution
• Describe the process of codec negotiation in SIP, the related protocols, and the Early Offer mechanism
• Describe the commands that are used to configure basic SIP functionality on Cisco IOS gateways
• Explain the major SIP features that support ISDN and describe how to configure them
• Explain how to configure SIP support for SRTP
• Describe how to configure SIP interface binding, transport protocol, and SIP timers
• Describe key commands that are used to monitor and verify SIP gateway operation

Lesson 4: Explaining MGCP Signaling Protocol

This lesson defines the characteristics of SIP, and explains when to use it. Upon completing this lesson, you will be able to meet these objectives:

• Describe MGCP, its components, and the advantages of MGCP as a voice gateway protocol
• Explain MGCP signaling messages and the interactions between an MGCP call agent and its associated gateways
• Describe the process of codec negotiation in MGCP and explain how DTMF digits are collected in MGCP
• Configure an MGCP residential and trunking gateway
• Describe how to configure the MGCP interface binding and other parameters to conform to the requirements of the call agent, trunks, or lines that are being used with the gateway
• Describe major commands that are used to verify an MGCP gateway

Lesson 5: Describing Requirements for VoIP Call Legs

This lesson defines special requirements for VoIP call legs, including the need for QoS, fax/modem relay and DTMF support. Upon completing this lesson, you will be able to meet these objectives:

• Describe the factors present in IP networks that affect audio clarity
• Describe the QoS requirement of VoIP and QoS features as they relate to a VoIP network
• Describe the challenge of transporting fax and modem calls over IP networks
• Explain how fax/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
• Explain DTMF relay, its options, and how DTMF relay is supported in MGCP, H.323, and SIP environments

Lesson 6: Describing Requirements for VoIP Call Legs

This lesson defines how to configure VoIP call legs in a gateway. Upon completing this lesson, you will be able to meet these objectives:

• Explain the key configuration components of a basic VoIP dial peer and describe how to configure VoIP dial peers
• Describe how to configure DTMF relay
• Explain how to configure fax/modem pass-through and relay
• Describe how to configure a single codec or codec negotiation on an SIP and H.323 gateway
• Explain how to limit the number of concurrent calls on a VoIP dial peer
• Lab 2-1: Configuring VoIP Call Legs

Module 3: Cisco Unified Communications Manager Express Endpoints Implementation

Describe how to implement IP phones using Cisco Unified Communications Manager Express.

Lesson 1: Introducing Cisco Unified Communications Manager Express

This lesson defines the functions and operation of the Cisco Unified Communications Manager Express. Upon completing this lesson, you will be able to meet these objectives:

• Describe the functions of the Cisco Unified Communications Manager Express in a voice network
• Identify the key features and benefits of Cisco Unified Communications Manager Express
• Describe the supported platforms and the required memory, licensing, and software that is needed to deploy Cisco Unified Communications Manager Express
• Explain the operation of the Cisco Unified Communications Manager Express when calls are made between the Cisco Unified Communications Manager Express and PSTN and between two Cisco Unified Communications Manager Express servers

Lesson 2: Examining Cisco Unified Communications Manager Express Endpoint Requirements

This lesson defines Describe all components that are required to support endpoints by Cisco Unified Communications Manager Express and explain how to configure them. Upon completing this lesson, you will be able to meet these objectives:

• Describe the Cisco Unified Communications Manager Express 8.0 SCCP and SIP endpoints and explain their capabilities
• Explain the Cisco Unified IP phone endpoint boot process and identify its major requirements (PoE, VLAN, DHCP, TFTP)
• Describe options to power endpoints and describe their characteristics
• Describe endpoint VLAN requirements; explain voice and data VLANs and how to configure VLANs to enable endpoint registration
• Identify DHCP service options and DHCP relay, and describe how to configure them to support Cisco Unified Communications Manager Express endpoints
• Describe NTP and how to configure it
• Describe Cisco Unified IP phone firmware files and XML configuration files, and identify how Cisco Unified IP phones obtain the files via TFTP
• Describe how to configure system-level parameters in an SCCP environment
• Describe how to set up system-level parameters in a SIP environment

Lesson 3: Configuring Cisco Unified Communications Manager Express Endpoints

This lesson defines Cisco Unified Communications Manager Express endpoint configuration elements such as phones and directory numbers. Upon completing this lesson, the learner will be able to meet these objectives:

• Describe the types of directory numbers and how they are implemented in Cisco Unified Communications Manager Express
• Explain how to configure directory numbers for SCCP phones
• Describe how to define an IP phone type by configuring an ephone-type template
• Explain how to configure major parameters of SCCP phones and how to assign directory numbers to the phones
• Explain how to configure directory numbers for SIP phones
• Explain how to configure major parameters of SIP phones and how to assign directory numbers to the SIP phones
• Explain how to enable Cisco IP Communicator to register with Cisco Unified Communications Manager Express
• Explain how to generate configuration files for SCCP and SIP endpoints and how to reset and restart SCCP and SIP phones
• Describe how to monitor and verify all major aspects of Cisco Unified Communications Manager Express endpoint operation
• Lab 3-1: Configure Cisco Unified Communications Manager Express to Support Endpoints

Module 4: Dial Plan Implementation

Describe the components of a dial plan and explain how to implement a dial plan on a Cisco Unified voice gateway.

Lesson 1: Introducing Call Routing

This lesson defines the characteristics and requirements of a numbering plan. Upon completing this lesson, you will be able to meet these objectives:

• Describe the basic characteristics of a typical numbering plan and list the different types of numbering plans
• Explain the attributes of a scalable numbering plan
• Describe overlapping numbering plans and strategies to address the issue of overlap
• Describe how to integrate private and public PSTN numbering plans
• Explain how a gateway implements the numbering plan
• Describe how a voice gateway routes calls

Lesson 2: Understanding Dial Plans

This lesson defines the components of a dial plan and their functions. This ability includes being able to meet these objectives:

• Describe the characteristics and components of a typical dial plan
• Explain the concept of endpoint addressing, including overlapping directory numbers
• Describe the characteristics of call routing and the importance of path selection
• Explain PSTN dial plan requirements
• Describe special ISDN dial plan requirements
• Provide the characteristics of digit manipulation in a voice gateway implementation
• Describe the implementation of calling privileges in a voice gateway
• Describe the characteristics of call coverage in gateway implementation

Lesson 3: Describing Digit Manipulation

This lesson defines how to configure a gateway for digit manipulation. Upon completing this lesson, you will be able to meet these objectives:

• 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
• Lab 4-1: Implementing Digit Manipulation

Lesson 4: Configuring Path Selection

This lesson defines how to configure a gateway to perform path selection. Upon completing this lesson, the learner will be able to meet these objectives:

• Describe how the voice gateways select the correct path when routing voice calls
• Explain how a gateway matches dial peers to determine path selection
• Describe the various path selection strategies
• Describe the characteristics of site-code dialing and toll bypass
• Describe how to configure site-code dialing and toll bypass in a gateway
• Explain the principle and characteristics of TEHO
• Describe how to configure TEHO
• Lab 4-2: Implementing Path Selection

Lesson 5: Configuring Calling Priveleges

This lesson defines how to configure calling privileges on a voice gateway. Upon completing this lesson, the learner will be able to meet these objectives:

• Describe calling privileges characteristics and explain their operations
• Describe how to implement calling privileges on Cisco IOS gateways
• Describe how to implement calling privileges in Cisco Unified SRST and Cisco United Communications Manager Express and how it differs from the implementation on voice gateways
• Describe how to configure COR
• Describe how to verify COR
• Lab 4-3: Implementing Calling Privileges

Module 5: Gatekeeper and Cisco Unified Border Element Implementation

Explain what gatekeepers and Cisco Unified Border Elements are, how they work, and what features they support.

Lesson 1: Understanding Gatekeepers

This lesson describes Cisco gatekeeper functions and configuration. Upon completing this lesson, you will be able to meet these objectives:

• Describe the functionality of gatekeepers in an H.323 environment
• Describe the signaling between gateways and gatekeepers
• Explain the gatekeeper call routing process, and the related elements, such as gatekeeper zone, zone prefixes, technology prefixes, and E.164 aliases
• Describe how a gatekeeper supports CAC functions
• List the steps necessary to configure a multizone gatekeeper for local and remote zone call routing
• Describe how to configure local and remote zones on a gatekeeper
• Explain how to configure gatekeeper zone prefixes
• Describe how to configure gatekeeper technology prefixes
• Explain how to adapt configuration of an H.323 gateway to register with a Gatekeeper
• Describe how to configure CAC functions on a gatekeeper
• Explain how to verify that H.323 endpoints are registered properly and calls are correctly routed across a gatekeeper
• Lab 5-1: Implementing Gatekeepers

Lesson 2: Examining Cisco Unified Border Element

This lesson defines Cisco Unified Border Element features and configuration. Upon completing this lesson, you will be able to meet these objectives:

• 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
• Lab 5-1 Configuring Basic Gatekeeper Functionality

Lesson 3: Implementing Gatekeeper-Based CAC

This lesson defines how to implement gatekeeper-based CAC using zone bandwidth. Upon completing this lesson, you will be able to meet these objectives:

• Describe bandwidth operation in a gatekeeper zone
• Describe the functionality of a Cisco Unified Border Element and its applications in enterprise VoIP environments
• Explain how protocol interworking is performed on a Cisco Unified Border Element and what interworking options are supported
• Describe how media flows are managed by a Cisco Unified Border Element
• Explain how to verify Cisco Unified Border Element operation
• Explain how Cisco Unified Border Element can be used to perform RSVP-based CAC
• Describe call flows in typical Cisco Unified Border Element deployments
• Explain how to configure H.323-to-H.323 interworking on a Cisco Unified Border Element
• Describe how to configure basic H.323-to-SIP interworking on a Cisco Unified Border Element, including DTMF relay interworking
• List the commands that are used to configure media flow-around, media flow-through, and transparent codec pass-through
• Lab 5-2: Implementing Cisco Unified Border Element

Module 6: Quality of Service

Describe why QoS is needed, what functions it performs, and how it can be implemented in a Cisco Unified Communications network.

Lesson 1: Introducing QoS

This lesson defines Explain the functions, goals, and implementation models of QoS, and what specific issues and requirements exist in a converged Cisco Unified Communications network. Upon completing this lesson, you will be able to meet these objectives:

• Explain the four key quality issues for voice traffic that exist in Cisco Unified Communications networks and describe how they impact voice quality
• Define QoS goals with respect to voice traffic
• Explain the three key steps that are involved in implementing a QoS policy in a Cisco Unified Communications network
• Describe how traffic is identified and divided into classes, and how QoS policies are defined for the traffic classes
• List four methods for implementing and managing a QoS policy-CLI, MQC, Cisco AutoQoS, and QPM-and describe their characteristics
• Describe briefly the three key models for providing QoS in a network

Lesson 2: Understanding QoS Mechanisms and Models

This lesson defines the characteristics and QoS mechanisms of the DiffServ model and contrasts it to other models. Upon completing this lesson, you will be able to meet these objectives:

• Explain the purpose and function of DiffServ
• Describe the basic format and the purpose of the DSCP field in the IP header and contrast it to the traditionally used IP precedence format
• List the different per-hop behaviors that are used in DSCP
• Describe the interoperability between DSCP-based and IP precedence-based devices
• Explain the key mechanisms of DiffServ to implement QoS in an IP network
• Describe Cisco QoS baseline model for enterprise

Lesson 3: Explaining Classification, Marking, and Link Efficiency Mechanisms

This lesson defines Explain the operation and configuration of the QoS classification and marking mechanisms, including the concept of trust boundaries and describes how LFI and cRTP provide link efficiency on WAN links and how they are configured. Upon completing this lesson, you will be able to meet these objectives:

• Describe the three steps that are involved in implementing a QoS policy using MQC, and the differences between class maps, policy maps, and service policies
• Describe how a class map is used to define a class of traffic and list which classification options exist and explain the purpose of packet marking and describe how a policy map is used to implement traffic marking policy
• Identify the Cisco IOS commands that are used to configure and monitor classification and class-based marking
• Explain QoS trust boundaries and explain their significance in LAN-based classification and marking
• Describe data link layer-to-network layer interoperability between different QoS markers and explain how to configure the mapping
• Explain the purpose of the link efficiency mechanisms and their functions
• Define link categories and explain when link efficiency mechanisms are mandatory
• Explain VoIP susceptibility to increased latency when large packets such as FTP transfers traverse slow WAN links, and specify what serialization delays are generally acceptable for voice
• Describe LFI and explain how to calculate recommended fragment size
• Describe how to configure and monitor MLP LFI
• Explain how to configure and monitor FRF.12
• Explain how to configure and monitor cRTP

Lesson 4: Managing Congestion and Rate Limiting

This lesson defines policing, shaping, and LLQ, their operations and configuration, using the MQC. Upon completing this lesson, you will be able to meet these objectives:

• Describe congestion, its origin, and the need for congestion management and rate limiting mechanisms
• Contrast the features of traffic policing and traffic shaping and identify the points in a network where policing and shaping is most effective
• Describe how a token bucket is used to measure traffic rates and explain token replenishment and consumption
• Explain the three models of class-based policing: single token bucket class-based policing, dual token bucket class-based policing, and dual-rate token bucket class-based policing
• Describe how to configure and monitor class-based policing
• Describe class-based shaping and explain the two shaping approaches: average rate and peak rate
• Describe how to configure and monitor class-based shaping
• Describe LLQ architecture, features, and operation
• Identify the MQC commands that are required to configure and monitor LLQ on a Cisco router
• Describe how to calculate voice bandwidth requirements for LLQ configuration across major data link layer technologies

Lesson 5: Understanding Cisco AutoQoS

This lesson defines how AutoQoS works and what it achieves in a Cisco Unified Communications network. Upon completing this lesson, the learner will be able to meet these objectives:

• Explain how Cisco AutoQoS VoIP is used to implement QoS policy and identify the router and switch platforms on which Cisco AutoQoS VoIP can be used
• Describe the prerequisites for Cisco AutoQoS VoIP and its configuration using the CLI
• Explain how to examine and monitor a network configuration after Cisco AutoQoS has been enabled
• Identify the QoS technologies that are automatically implemented on the network using Cisco AutoQoS VoIP
• Explain how Cisco AutoQoS for the Enterprise is used to implement QoS policy
• Describe how Cisco AutoQoS for the Enterprise is configured on a router using the CLI
• Describe how to examine and monitor a network configuration after Cisco AutoQoS for the Enterprise has been enabled
• Lab 6-1: Implementing QoS Using AutoQos and Manual Configuration

Hands-on Lab Exercises

• Lab 1-1: Configuring Voice Ports
• Lab 1-2: Configuring DSPs
• Lab 2-1: Configuring VoIP Call Legs
• Lab 3-1: Configure Cisco Unified Communications Manager Express to Support Endpoints
• Lab 4-1: Implementing Digit Manipulation
• Lab 4-2: Implementing Path Selection
• Lab 4-3: Implementing Calling Privileges
• Lab 5-1: Implementing Gatekeepers
• Lab 5-2: Implementing Cisco Unified Border Element
• Lab 6-1: Implementing QoS Using AutoQoS and Manual Configuration