Voice over IP Foundations Bootcamp

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, Asterisk PBX, Kamailio SIP Proxy, Linksys Ethernet phone, and SIP-based ATA. You will also cover Cisco QoS policy administration and demonstrate successful VoIP calls in high data traffic conditions.

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Overview

  • 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

Target 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

Expected Duration

5 day

Course Objectives

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
    • Zero Packet Loss
    • Ten Percent Packet Loss
    • Twenty Percent Packet Loss
  • T.38 Fax Spoofing
    • Call Setup
    • Discovering the Fax Tone
    • T.30 Negotiation
    • Shifting to 9.6 Kbps
    • T.38 Phase

9. Real-Time Transport Protocol (RTP)

  • RTP Architecture
    • RTP and RTP Control Protocol
    • Encapsulating the Voice Packet
    • RTP Ports
  • RTP Profile
    • Payload Types
    • Mapping Payload Type to Codec Type
    • How H.323 Identifies the Payload Type
    • NTP vs. RTP Timestamp
    • RTP Timestamps
    • RTP Timestamps and Silence Suppression
    • RTP Timestamps and Jitter Calculation
  • Controlling Jitter
    • Jitter Buffer Delay
  • Mixers
    • Synchronization Source
    • Conference Bridge Adds CSRC
  • RTP Header
    • UDP Packet with RTP Header and Voice
    • Required Fields
    • Version
    • Padding Bit
    • Extension Bit
    • CSRC
    • Market Bit
    • Payload Type
    • Sequence Number
    • Timestamp
    • SSRC
    • The Format-Specific Parameter (fmtp) Attribute
    • RFC 2833 Example: A Dialing Event
      • Transmitter Processing
      • Receiver Processing
  • Controlling Serialization Delay
    • Perfect Candidate for LFI and RTP Header Compression
  • RTP Header Compression Process (RFC 2508)
    • RTP Header Compression Format
  • RTCP
    • RTCP QoS: Round-Trip Delay Calculation
    • Sender Reports
    • Receiver Reports
    • Source Descriptions
    • Source Description Items
    • Other RTCP Packets

10. SIP Architecture

  • SIP User Agents
    • SIP Requests (Methods)
    • SIP Response Codes
  • SIP Proxy
    • SIP Back-to-Back UA
    • Session Border Controller
    • Forking Proxy
    • SIP Redirect Proxy
  • Global SIP Architecture
    • Overview of Operation
    • Classic SIP Trapezoid
    • INVITE Request
    • Session Description Protocol
    • Proxy Function
    • 180 Response
    • 200 Final Response
    • BYE
    • INVITE and ACK
  • SIP Functional Stack
  • SIP Core Documents and Extensions

11. SIP Call Flow Examples

  • SIP Call Analysis
    • SIP Registration with Authentication
    • SIP Call without INVITE Authentication
    • The 100rel Process
    • Busy Number
    • Abandoned Call (Cancel)
    • SIP Redirect (Call Forward)
    • Call Transfer
  • E&M Tie Trunk
    • See a Problem?
    • Solution: SIP 183 Response

12. Session Description Protocol

  • Session Description Protocol
    • v= Header
    • o= Header
    • s= Header
    • c= Header
    • t= Header
    • m= Header
    • a= Header
  • Offer/Answer Model
    • Offer/Answer: Example 1
    • Offer/Answer: Example 2
    • SDP Offer/Answer Rules
    • UPDATE Method
    • RTP SEND and RECV Defined
    • Media Direction and RTCP
    • How RTCP Works
    • Placing a Call on HOLD

13. SIP NAT Traversal

  • SIP NAT Traversal
    • One-Way Voice Results
    • Full Cone NAT
    • IP Address Restricted NAT
    • Port Restricted NAT
    • Symmetric NAT
    • Simple Traversal of UDP through NATs
    • Traversal Using Relay NAT
    • NAT with Embedded SIP Proxy
    • Public VoIP Example

14. Media Gateway Control Protocol (MGCP)

  • Protocol Comparison
  • MGCP Call Model
    • Hairpin Call Example
    • Defined Endpoints
  • MGCP Commands
    • MGCP Syntax Example
    • Return Codes
    • Return Code Table
    • Parameter Lines
    • DTMF Package
    • Line Package
  • Digit Maps
  • MGCP Trace Procedure
    • MGCP Trace (Steps 1-8)
    • MGCP Trace (Steps 9-14)
    • MGCP Trace (Steps 15-22)
    • MGCP Trace (Steps 23-28)
  • MGCP Established Call
    • MGCP Trace (Steps 29-36)
    • MGCP Trace (Steps 37-40)

15. Queuing

  • CoS vs. QoS
    • Leaky Bucket
    • First In, First Out
    • Type Classification
    • Session ID Classification (Fair Queuing)
    • Dequeuing

16. QoS-Related Protocol

  • Sources of Delay
    • Packetization Delay
    • Algorithmic Delay (Look Ahead)
    • Coder Processing Delay (Think Time)
    • Queuing Delay
    • Serialization Delay
  • Low-Speed Link
    • How 56-Kbps Links Cause Jitter
    • Upgrade to T1/E1 and Prioritize Voice
  • QoS Technology Solutions: Differentiated Services (DiffServ)
    • Supporting a VoIP Call with DiffServ
    • ToS Field
    • DiffServ Process at the Edge Router
    • DiffServ Process in the Core
    • DiffServ Highlights
  • Traffic Engineering: An Art Form
    • Measuring Engineering
    • Grade of Service

Appendix A: Glossary

Appendix B: H.323

Labs

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