OneSource Professional Training Solutions
OneSource Professional Training Solutions, Inc. presents

Voice over IP Foundations On-Site Training

This on-site training class is also available as Public Schedule Seminar.

Voice over IP Foundations

Course Description/Agenda

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 and hardware, such as Wireshark, trixbox (formerly Asterisk@Home), Linksys Ethernet phone, SIP-based ATA, and SIP-based Server and PBX products from Brekeke Software, Inc.

What You'll Learn

  • Core concepts of how Internet Protocol (IP) carries a VoIP packet
  • Advantages and disadvantages of SIP Trunking
  • 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)
  • SIP proxy, Session Border Controller (SBC), and SIP softswitch
  • Media Gateway Control Protocol (MGCP) analysis
  • MGCP architecture
  • 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, and MGCP call flows
  • Configure the trixbox Softswitch and SIP proxy
  • Configure SIP gateways and softphones

Who Should 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.

Course Outline

1. Packetizing Voice

  • Telephony Architecture
    • Introduction to the VoIP Standards
  • Connecting VoIP to PSTN
    • Traffic Engineering
    • PSTN to VoIP Using Magic
  • Voice Digitization
    • Companding Mu-Law vs. A-Law
  • Time Division Circuit Switching
  • Voice Packet
    • The 20-Millisecond Voice Packet
    • The 60-Millisecond Voice Packet
    • The Voice Packet Header
    • Other Voice Packet Sample Sizes
    • Voice Packet Analysis
    • Voice Packet Analysis: Other Voice Packet Sample Sizes
  • QoS Overview
    • Latency
    • Packet Loss
    • Jitter
  • Controlling Delay
    • Sources of Delay
    • The First Voice Packet
    • The Second Voice Packet
    • The Third Voice Packet
    • Jitter Buffer Under Perfect Conditions
    • An Adaptive Jitter Buffer

2. SIP Trunking

  • The Legacy Circuit Switch
  • VoIP Phases
    • VoIP Phase 1: LAN Connect the Line Side
    • VoIP Phase 2: Decompose the Switch Cabinet
    • VoIP Phase 3: Shrink the MGs and Add Survivability
    • VoIP Phase 4: Add SIP Trunking
    • VoIP Phase 5: Eliminate the Old MGs
    • VoIP Phase 6: Add EMUN
    • VoIP Phase 7: Mass Acceptance of SIP Trunking with ENUM?
  • SIP Trunking Costs
  • Other Means of Connection
  • The "Old PBXcan do SIP Trunking if the Vendor Offers the Software
  • SIP Trunking Protocols
    • Peer-to-Peer RTP
    • Hairpin RTP
  • Disadvantages and Advantages of SIP Trunking
    • Disadvantages
    • Advantages
  • ITSPs
  • SIP Trunking Examples
    • SIP Trunk Outbound Call
    • Public VoIP

3. VoIP in the LAN

  • IP and Ethernet
    • A Sample Ethernet Switched Network
  • MAC Addresses
  • IP MAC Address Learning
    • Unknown Destination MAC Addresses
    • Flood the Broadcast
    • Response to Flooded Packet
    • Learning Port Information
    • Switching
  • MAC Table Aging
  • Ethernet Communications Limits
  • Virtual LANs
    • VLAN Trunk
    • VLAN Tags
    • Untagged Frames
  • Port-Based VLANs
    • Broadcast Frame in VLAN 10
  • VLAN Trunking for VoIP Phones
  • IEEE 802.3af Device Detection
    • IEEE 802.3af Power Classifications
    • QoS at Layer 2
    • VLAN Tagging Process
    • IEEE 802.1q Frame Tagging

4. IP Networking

  • One-Way vs. Both-Way Routing
  • Static Routing
    • Subnet Masks and Routing
    • Routing and Switching
  • Routing Protocols
    • Distance Vector Routing
    • Link-State Routing

5. TCP/IP Review

  • Transmission Control Protocol (TCP) vs. User Datagram Protocol (UDP)
    • Connection-Oriented Protocol (TCP)
    • TCP/IP Packet Format and Operation
    • Connectionless Protocols (UDP)
    • UDP Packet
  • DNS
    • Basic Method of DNS

6. Dial Plan Essentials

  • Dial Plan Example
  • Digit Map
  • Enbloc vs. Overlap
  • Common Modifications to REGEX
  • Symbols
    • Regular Expressions
    • Metacharacters
  • Matching
  • Normalization Examples

7. SIP-Related IP Services

  • DHCP Option for SIP
    • DHCP Discover
    • DHCP Offer
  • Root-Level Domain Registration
  • Basic Method of DNS
    • Why Start with ENUM?
  • ENUM: NAPTR Query
    • ENUM: NAPTR Response
  • Locating SIP Servers: An Example
    • NAPTR Response
    • SRV Query
    • SRV Response
    • A Record Query
  • Regular Expressions
    • The Metacharacters

8. Voice Compression

  • Voice Compression Hardware
    • ASICs
    • DSPs
  • Mean Opinion Scores
  • Codecs
    • G.711, G.723.1, G.726
    • G.728 and G.729
  • Voice Compression
    • Formants
    • The Predictor
    • PCM Sampling
  • Voice Compression Algorithms
    • ADPCM Compression
    • Vocoder
    • G.729 Example
  • Codec Comparison Exercise
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