IP Telephony v1.0 Scope and Sequence. Cisco Networking Academy Program

Transcription

1 IP Telephony v1.0 Scope and Sequence Cisco Networking Academy Program

2 Table of Content COURSE OVERVIEW...4 Course Description...4 Course Objectives...4 Target Audience...5 Prerequisites...5 Lab Requirements...5 Certification Information...5 Course Duration...5 IP TELEPHONY V1.0 OUTLINES...6 IP Telephony Lab Outlines...6 Introduction to Packet Voice Technologies... 6 Introduction to VoIP... 6 Configuring Cisco CallManager Express... 6 Voice Dial Plans, Configuring Voice Interfaces and Dial Peers... 6 Configuring CME Additional Features... 6 VoIP Signaling and Call Control... 7 Improving and Maintaining Voice Quality... 7 Job Aids... 7 IP Telephony Content Resource Outlines...8 Introduction to Packet Voice Technologies... 8 Traditional Telephony... 8 Packetized Telephony Networks... 8 IP Telephony Applications... 8 Analog Voice Basics... 8 Analog-to-Digital Voice Encoding... 8 Signaling Systems... 9 Introduction to VoIP Requirements of Voice in an IP Internetwork Gateway and Their Roles Encapsulating Voice in IP Packets Calculating Bandwidth Requirements Configuring Cisco CallManager Express (CME) Overview of Cisco CME Differences between Traditional Telephony and VoIP Challenges and Solutions in VoIP Cisco CME Features and Functionality Cisco CME Network Parameters IP Phone Registration Ephone-dn and Ephone CallManager Express Files Initial Phone Setup Voice Dial Plans, Configuring Voice Interfaces and Dial Peers Call Establishment Principles Configuring Dial Peers Special-Purpose Connections Building a Scalable Numbering Plan Configuring Voice Ports Adjusting Voice Quality Analog and Digital Voice Interfaces Configuring Analog and Digital Voice Interfaces Dial Peers Call Setup and Digit Manipulation Class of Restriction Configuring CME Additional Features Cisco CME GUI Features Configuring Phone Features VoIP Signaling and Call Control Need for Signaling and Call Control Configuring H Copyright 2005, Cisco Systems, Inc. Scope and Sequence > Table of Content 2

3 Configuring MGCP Improving and Maintaining Voice Quality IP QoS Mechanisms Implementing AutoQoS Comparing Voice Quality Measurement Standards VoIP Challenges QoS and Good Design Jitter Delay Apply QoS in the Campus QoS Tools in the WAN Configuring QoS in the WAN Configuring CAC Voice Bandwidth Engineering Copyright 2005, Cisco Systems, Inc. Scope and Sequence > Table of Content 3

4 Course Overview IP Telephony is the most requested new course offering within the Academy community. To meet the demand, Cisco has created an R&D course built on suggestions from Academy Instructors. Course content will focus on lab activities surrounding voice and data convergence, but does NOT include traditional on-line curriculum or assessment. IP Telephony will focus on entry level skills required to implement IP Telephony in a SOHO environment. Course Description The Cisco IP Telephony version 1.0 course provides an introduction to converged voice and data networks as well as the challenges faced by its various technologies. The course presents Cisco solutions and implementation considerations to address those challenges. In this course, students will learn about Cisco CallManager Express (CME) architecture, components, functionality and features. They will also learn some Voice over IP (VoIP) and Quality of Service (QoS) technologies and apply them to Cisco CME environment. The focus of the course is: Course Objectives Call Manager Express (Windows-based call manager is not taught at this time) Connecting to a PSTN network Connecting from one router across a WAN to another router running CME Connecting from one CME enabled router to another CME enabled router Upon completing this course, the learner will be able to meet these overall objectives: Determine the relevant critical business and technical needs in order to develop a Cisco IP Telephony design framework in a SOHO environment, including the choice of signaling type and encoding methods. Select the appropriate hardware and software components to support a proposed SOHO IP telephony design Determine which high availability issues would influence the selection of network hardware and software for an SOHO environment. Design the appropriate dial plan to support SOHO design requirements Describe the similarities and differences between PSTN and VoIP including call transport, call signaling and bandwidth requirements Describe the technologies used in Voice over IP and how they differ from PSTN technologies. Determine how PSTN connectivity issues influence SOHO hardware and software selection. Copyright 2005, Cisco Systems, Inc. Scope and Sequence > Course Overview 4

5 Target Audience Understand and be able to configure various connection types to the PSTN. Explain the benefits of hierarchical and scalable dial and numbering plans Install an IP Telephony solution in a SOHO environment using CME Configure Cisco Call Manager Express to support IP Phones Secure an IP Communications network Describe effective troubleshooting methods to resolve issues in SOHO IP Telephony networks Use appropriate troubleshooting methods to determine and solve QoS issues in an SOHO IP Telephony networks The target audience is individuals desiring to continue their post-ccna preparation in designing, maintaining and troubleshooting an SMB IP telephony network. Prerequisites Prior to taking this course, students should have completed CCNA 1 through 4 or the equivalent. The following prerequisites are beneficial, but not required: CCNA certification Work experience Lab Requirements Please refer to the IP Telephony Equipment Bundle Spreadsheets on Cisco Academy Connection (CAC). Certification Information IP Telephony version 1.0 does not align to any one Cisco certification. Course Duration The course is designed to be delivered in a 40 contact hour time frame. Approximately 30 hours will be devoted to lab activities and a hands-on skills exam and 10 hours will be spent on curriculum content. Case studies on IP telephony are recommended, but format and timing are to be determined by the Local Academy. Copyright 2005, Cisco Systems, Inc. Scope and Sequence > Course Overview 5

6 IP Telephony v1.0 Outlines IP Telephony Lab Outlines Introduction to Packet Voice Technologies There is no lab in this module. Introduction to VoIP Basic Setup for the CME Router with Switch Module Basic Setup for the CME Router and Switch Installing Cisco CME Software Connecting the IP Phone to a Switch Resetting a 7900 Series Cisco IP Phone to Factory Defaults Configuring Cisco CallManager Express CME Automated Phone Setup CME Manual Phone Setup CME Partially Automated Phone Setup Voice Dial Plans, Configuring Voice Interfaces and Dial Peers Configuring a FXS Port Configuring a FXO Port Configuring PRI Interface and DID Configuring VoIP Dial-Peers Across a WAN Link Configuring Class of Restriction Configuring CME Additional Features Configure GUI for System Administrator Configure GUI for Customer Administrator Configure GUI for Phone User Configuring Call Transfer and Call Forwarding Configuring Call Park Customize the IP Phone Display Copyright 2005, Cisco Systems, Inc. Scope and Sequence > IP Telephony Lab Outlines 6

7 Configure the Intercom Feature Configuring a Dialable Intercom Configure Paging Group VoIP Signaling and Call Control There is no lab in this module. Improving and Maintaining Voice Quality Configuring AutoQoS Job Aids Table 1 IP Telephony Addressing Scheme Table 2 IP Telephony Dial Plan TCP and UDP Port Numbers Adtran Atlas 550 Configuration Copyright 2005, Cisco Systems, Inc. Scope and Sequence > IP Telephony Lab Outlines 7

8 IP Telephony Content Resource Outlines Introduction to Packet Voice Technologies Traditional Telephony Basic Components of a Telephony Network CO Switches Private Switching Systems Call Signaling Multiplexing Techniques Packetized Telephony Networks Benefits of Packet Telephony Networks Call Control Distributed vs. Centralized Call Control Packet Telephony Components Best-Effort Delivery of Real-Time Traffic IP Telephony Applications Analog Interfaces Digital Interfaces IP Phones Analog Voice Basics Local-Loop Connections Types of Local-Loop Signaling Supervisory Signaling Address Signaling Informational Signaling Trunk Connections Types of Trunk Signaling E&M Signaling Types Trunk Signal Types Used by E&M Line Quality Management of Echo Analog-to-Digital Voice Encoding Basic Voice Encoding: Converting Analog to Digital Basic Voice Encoding: Converting Digital to Analog Copyright 2005, Cisco Systems, Inc. Scope and Sequence > IP Telephony Content Resource Outlines 8

9 The Nyquist Theorem Voice Compression and Codec Standards Compression Bandwidth Requirements Voice Quality Measurement Signaling Systems CAS Systems: T1 CAS Systems: E1 CCS Systems ISDN Copyright 2005, Cisco Systems, Inc. Scope and Sequence > IP Telephony Content Resource Outlines 9

10 Introduction to VoIP Requirements of Voice in an IP Internetwork Real-Time Voice in a Best-Effort IP Internetwork Packet Loss, Delay, and Jitter Consistent Throughput Reordering of Voice Packets Reliability and Availability Gateway and Their Roles Understanding Gateways Guidelines for Selecting the Correct Gateway Determining Gateway Interconnection Requirements in an Enterprise Environment, Central and Remote Site Determining Gateway Interconnection Requirements in a Service Provider Environment Encapsulating Voice in IP Packets Major VoIP Protocols RTP and RTCP Reducing Header Overhead with CRTP When to Use RTP Header Compression Calculating Bandwidth Requirements Codec Bandwidths Impact of Voice Samples and Packet Size on Bandwidth Data Link Overhead Security and Tunneling Overhead Specialized Encapsulations Calculating the Total Bandwidth for a VoIP Call Effects of VAD on Bandwidth Copyright 2005, Cisco Systems, Inc. Scope and Sequence > IP Telephony Content Resource Outlines 10

11 Configuring Cisco CallManager Express (CME) Overview of Cisco CME What is Cisco CallManager Express? How Does Cisco CallManager Express Work? Licensing Differences between Traditional Telephony and VoIP Traditional Telephony PCM Theory Basic Voice Encoding: Converting Digital to Analog PCM Theory Coder-Decoder Encapsulating Voice in IP Packets Challenges and Solutions in VoIP Challenges in VoIP Bandwidth Requirements in VoIP Cisco CME Features and Functionality Supported Protocols and Integration Options Cisco CallManager Express Requirements Cisco CallManager Express Restrictions Cisco CME Network Parameters Auxiliary VLANs Configuring Auxiliary VLANs DHCP Service Setup IP Phone Registration Files IP Phone Information Download and Registration Ephone-dn and Ephone Ephone-dn Ephone Type of Ephone-dns Number of Ephone-dns Copyright 2005, Cisco Systems, Inc. Scope and Sequence > IP Telephony Content Resource Outlines 11

12 CallManager Express Files Cisco CME File Bundled Cisco CME File Individual Cisco CME Files GUI Files Cisco CME - TAPI Integration Additional Files Initial Phone Setup Setting Up Phones in a Cisco CME System Automated Phone Setup Partially Automated Phone Setup Manual Phone Setup Setup Tips Optional Parameters Router Configuration: Two Commands Optional Parameters Locale Parameters Rebooting Cisco CallManager Express Phones Setup Troubleshooting Verifying Cisco CallManager Express Phone Configuration Copyright 2005, Cisco Systems, Inc. Scope and Sequence > IP Telephony Content Resource Outlines 12

13 Voice Dial Plans, Configuring Voice Interfaces and Dial Peers Call Establishment Principles What Are Call Legs? End-to-End Calls Configuring Dial Peers Understanding Dial Peers Configuring POTS Dial Peers Configuring VoIP Dial Peers Configuring Destination-Pattern Options Default Dial Peer Matching Inbound Dial Peers Matching Outbound Dial Peers Hunt-Group Commands Configuring Hunt Groups Digit Collection and Consumption Understanding Digit Manipulation Practice Item Answer Key Special-Purpose Connections Connection Commands PLAR and PLAR-OPX Configuring Trunk Connections Tie-Line Connections Building a Scalable Numbering Plan Scalable Numbering Plan Scalable Numbering Plan Attributes Hierarchical Numbering Plans Internal Numbering and Public Numbering Plan Integration Enhancing and Extending an Existing Plan to Accommodate VoIP Configuring Voice Ports Voice Port Applications FXS Ports FXO Ports E&M Ports Timers and Timing Copyright 2005, Cisco Systems, Inc. Scope and Sequence > IP Telephony Content Resource Outlines 13

14 Digital Voice Ports ISDN CCS Options Monitoring and Troubleshooting Adjusting Voice Quality Electrical Characteristics Voice Quality Tuning Echo Cancellation Commands Analog and Digital Voice Interfaces Local-Loop Connections Analog Voice Interfaces Channel Associated Signaling Systems: T1 Channel Associated Signaling Systems: E1 Common Channel Signaling Systems PRI/BRI Configuring Analog and Digital Voice Interfaces Foreign Exchange Station Ports (FXS) Foreign Exchange Office Ports (FXO) Ear and Mouth Ports (E&M) Common Channel Signaling (CCS): ISDN BRI Timers and Timing Digital Voice Ports Channel Associated Signaling (CAS) Common Channel Signaling (CCS): ISDN Primary Rate Interface (PRI) Dial Peers What is Dial Peer? Plain Old Telephone Service Dial Peers VoIP Dial Peers Destination-Pattern Options What is the Default Dial Peer? Call Setup and Digit Manipulation End-to-End Calls Matching Inbound Dial Peers Matching Outbound Dial Peers Copyright 2005, Cisco Systems, Inc. Scope and Sequence > IP Telephony Content Resource Outlines 14

15 Digit Collection and Consumption What Is Digit Manipulation? PLAR Class of Restriction Class of Restriction (COR) Steps to Configure Class of Restriction Copyright 2005, Cisco Systems, Inc. Scope and Sequence > IP Telephony Content Resource Outlines 15

16 Configuring CME Additional Features Cisco CME GUI Features User Classes Cisco CallManager Express GUI Prerequisites Accessing the GUI Configuring Administrative User Classes Configuring Phone Features Call Transfer Call Forwarding IP Phone Display Calling and Directory Features Productivity Tools Custom IP Phone Rings Music on Hold Copyright 2005, Cisco Systems, Inc. Scope and Sequence > IP Telephony Content Resource Outlines 16

17 VoIP Signaling and Call Control Need for Signaling and Call Control VoIP Signaling Call Control Models Translation between Signaling and Call Control Models Call Setup Call Administration and Accounting Call Status and Call Detail Records Address Management Admission Control Centralized Call Control Distributed Call Control Centralized Call Control vs. Distributed Call Control Configuring H.323 H.323 and Associated Recommendations Functional Components of H.323 H.323 Call Establishment and Maintenance Call Flows without a Gatekeeper Call Flows with a Gatekeeper Multipoint Conferences Call Flows with Multiple Gatekeepers Survivability Strategies H.323 Proxy Server Cisco Implementation of H.323 Configuring H.323 Gateways Configuring H.323 Gatekeepers Monitoring and Troubleshooting Configuring MGCP MGCP and Its Associated Standards Basic MGCP Components MGCP Endpoints MGCP Gateways MGCP Call Agents Basic MGCP Concepts Copyright 2005, Cisco Systems, Inc. Scope and Sequence > IP Telephony Content Resource Outlines 17

18 MGCP Calls and Connections MGCP Events and Signals MGCP Packages MGCP Digit Maps MGCP Control Commands Call Flows Survivability Strategies Cisco Implementation of MGCP Understanding Basics of Cisco CallManager Configuring MGCP Monitoring and Troubleshooting MGCP Copyright 2005, Cisco Systems, Inc. Scope and Sequence > IP Telephony Content Resource Outlines 18

19 Improving and Maintaining Voice Quality IP QoS Mechanisms QoS Mechanisms Classification Marking Trust Boundaries Congestion Management Traffic Shaping Compression Link Fragmentation and Interleaving Implementing AutoQoS AutoQoS AutoQoS: Router Platforms AutoQoS: Switch Platforms AutoQoS Prerequisites Configuring AutoQoS Monitoring AutoQoS Automation with Cisco AutoQoS Comparing Voice Quality Measurement Standards Audio Clarity VoIP Challenges IP Networking Overview Jitter Delay QoS and Good Design Need for QoS Mechanisms Objectives of QoS Applying QoS for End-to-End Improvement of Voice Quality Jitter Understanding Jitter Overcoming Jitter Adjusting Playout Delay Parameters Symptoms of Jitter on a Network Dynamic Jitter Buffer Copyright 2005, Cisco Systems, Inc. Scope and Sequence > IP Telephony Content Resource Outlines 19

20 Static Jitter Buffer Delay Need for a Delay Budget Guidelines for Acceptable Delay Sources of Delay Effects of Coders and Voice Sampling on Delay Managing Serialization Delay Managing Queuing Delay Verifying End-to-End Delay Apply QoS in the Campus Need for QoS in the Campus Marking Control and Management Traffic QoS Tools in the WAN Need for QoS on WAN Links Recommendations for Generic QoS in the WAN Bandwidth Provisioning Optimized Queuing Link Efficiency Link Fragmentation and Interleaving CAC Configuring QoS in the WAN Configuring AutoQoS Configuring CAC Need for CAC CAC as Part of Call Control Services RSVP Understanding CAC Tools H.323 CAC Voice Bandwidth Engineering Erlangs Copyright 2005, Cisco Systems, Inc. Scope and Sequence > IP Telephony Content Resource Outlines 20