ECI – EMSP VOIP Course

Course Description

Gain essential data networking and Voice over IP (VoIP) knowledge in a single, week log 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 sides that illustrate the subject matter– text-only sides are kept to a minimum. In the skills-building labs, you will gain proficiency with some of the most popular VoIP software and hardware, such as Wireshark, trixbox (formerly Asterisk@Home)), Linksys Ethermet phone, SIP-based ATA, and SIP-based Server and PBX products and more.

What You’ll Learn in Class

• Core concepts of how Internet Protocol IP carries a VoIP packet
• Advantages and disadvantages of SIP Trunking
• Configure DHCP ad 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 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

Who Needs to Attend

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

Knowledge of TCP/IP networking, telecom architectures, phone systems,

Information Technologies

Hands-On 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: LoIP Island Configuration

Lab 10: SIP Ethernet Phone Configuration

Lab 11: Networking SIP Proxies

Lab 12: Dial Plan Implementation

Lab 13: SIP Softphone Configuration

Lab 14: Capturing and Analyzing RTP using Wireshark

Lab 15: Code MOS Testing

Lab 16: Increasing Packet Intervals

Lab 17: Codec bandwidth Testing

Lab 18: Silence Suppression

Lab 19: Codec Negotiation (Offer/Answer)

Lab 20: DTMF RFC 2833 and SIP INFO

LAB 2l: Using Wireshark for Capture and analysis

Lab 22: Sip REGISTER Authentication

Lab 23: SIP INVITE authentication

Lab 24: SIP Call Flow Analysis

Lab 25: Wi-Fi Radio Configuration

Lab 26: Wi-Fi Sip Phone Configuration

Lab 27: SIP Trunking

Lab 28: trixbox Meet-Me Conferencing

Lab 29: trixbox Voice Mail

Lab 30: QoS performance Testing

Lab 31: VoIP Gateway DiffServ Configuration

Lab 32: Queuing Strategies and QoS Configuration

Course Content

1. Packetizing Voice
2. SIP Trunking
3. VoIP on the LAN
4. IP Networking
5. TCP/IP Review
6. SIP-Related IP Services
7. Voice Compression
8. Real-Time Transport Protocol (RTP)
9. SIP Architecture
10. SIP Call Flow Examples
11. SIP Syntax
12. Session Description Protocol
13. SIP NAT Traversal
14. Media Gateway Control Protocol (MGCP)
15. H.323
16. Queuing
17. Qos-Related Protocol