Voice over IP Foundations

Convergent Communications, Inc. - A Cisco Systems Learning Partner

VOICE OVER IP FOUNDATIONS

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Voice over IP Foundations

Course Description

Gain essential data networking and Voice over IP (VoIP) knowledge in a single, week-long class. In this course, you will learn how VoIP works, why VoIP works, and how to use VoIP. On the first day, you will configure an IP network using Cisco routers and switches, learning IP fundamentals in order to make VoIP easier to understand. The remaining four days will focus on VoIP and IP telephony. The course is 60% hands-on labs and 40% lecture. The lecture portion of the class uses technically detailed slides that illustrate the subject matter - text-only slides are kept to a minimum. In the skills-building labs, you will gain proficiency with some of the most popular VoIP software, such as Wireshark, trixbox (formerly Asterisk@Home), Linksys Ethernet phone, SIP-based ATA, and SIP-based Server and PBX products from Brekeke Software, Inc.

This course includes 1-year access to our 50-book Online Networking Reference Library with titles specially selected to reinforce course concepts.

Course Objectives

Students will learn about :

Core concepts of how Internet Protocol (IP) carries a VoIP packet
Configure DHCP and DNS to support IP telephony
Real-Time Transport Protocol (RTP)
Session Initiation Protocol (SIP) - Call set up, Instant Messaging, Presence
Session Description Protocol (SDP)
The H.323 protocol suite, including H.225, RAS, and H.245
The role of endpoints, gatekeepers, gateways, and MCU in an H.323 network
SIP proxy, Session Border Controller (SBC), and SIP softswitch
Media Gateway Control Protocol (MGCP) analysis
MGCP architecture
A technical comparison of H.323, SIP, and MGCP
How to implement QoS to ensure the highest voice quality over your IP networks
The impact of jitter, latency, and packet loss on VoIP networks
How to use Wireshark to decode and troubleshoot RTP, SIP, MGCP, and H.323 call flows
Configure the trixbox Softswitch and SIP proxy
Configure SIP gateways and softphones
Security issues to consider when setting up VoIP

Intended Audience

This class is for people who need to understand VoIP technology. IT managers, technical sales/marketing personnel, consultants, network designers and engineers, product design engineers developing integrated-services products, telecom technicians and managers integrating PBX services within data networks, and systems administrators who will manage a converged network would benefit from this course.

Prerequisites

A Basic understanding of TCP/IP Networking.

Course Outline

Packetizing Voice
SIP Trunking
VoIP in the LAN
IP Networking
TCP/IP Review
SIP-Related IP Services
Voice Compression
Real-Time Transport Protocol (RTP)
SIP Architecture
SIP Call Flow Examples
SIP Syntax
Session Description Protocol
SIP NAT Traversal
Media Gateway Control Protocol (MGCP)
H.323
Queuing
QoS-Related Protocol

Course Labs

Lab 1: Network Hardware Installation
Lab 2: Cisco IOS Command Line Interface Configuration
Lab 3: Configure VLAN
Lab 4: IP Network Configuration
Lab 5: Implement DNS
Lab 6: Implement DHCP
Lab 7: Calling Without a SIP Proxy
Lab 8: UA Registration
Lab 9: SIP Ethernet Phone Configuration
Lab 10: SIP Server Configuration
Lab 11: Networking SIP Proxies
Lab 12: SIP Softphone Configuration
Lab 13: Wi-Fi Radio Configuration
Lab 14: Wi-Fi SIP Phone Configuration
Lab 15: Capturing and Analyzing RTP using Ethereal
Lab 16: Codec MOS Testing
Lab 17: Increasing Packet Intervals
Lab 18: Codec Bandwidth Testing
Lab 19: Silence Suppression
Lab 20: Codec Negotiation (Offer/Answer)
Lab 21: DTMF RFC 2833 and SIP INFO
Lab 22: Using Ethereal for Capture and Analysis
Lab 23: SIP REGISTER Authentication
Lab 24: SIP INVITE Authentication
Lab 25: SIP Call Flow Analysis
Lab 26: Trixbox Configuration
Lab 27: SIP Trunking
Lab 28: QoS Performance Testing
Lab 29: VoIP Gateway DiffServ Configuration
Lab 30: Queuing Strategies and QoS Configuration