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Voice over IP Foundations

Exclusive – Discover how and why Voice over IP works and gain an understanding of SIP.

GK# 3277

Course Overview

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In this lecture-only course, you will learn core concepts of how the Internet Protocol (IP) carries a Voice over IP (VoIP) packet. You will learn the fundamentals of Session Initiation Protocol (SIP) architecture, SIP-related IP services, the advantages and disadvantages of SIP Trunking as well as Quality of Service (QoS)-Related Protocol.

Schedule

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What You'll Learn

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  • Core concepts of how Internet Protocol (IP) carries a VoIP packet
  • Advantages and disadvantages of SIP Trunking
  • Understand how DHCP and DNS supports 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 Wireshark can decode and troubleshoot RTP, SIP, and MGCP call flows
  • Discuss trixbox Softswitch and SIP proxy
  • Discuss SIP gateways and softphones

Outline

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Viewing outline for:

Virtual Classroom Live Outline

Module 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

Module 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 PBX” can 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

Module 3 VoIP in the LAN

  • IP and Ethernet
  • A Sample Ethernet Switched Network
  • MAC Addresses
  • IP MAC Address Learning
  • known 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

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

Module 4 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

Module 5 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

Module 6 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

Module 7 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

Module 8 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

Module 9 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

Module 10 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

Module 11 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)

Module 12 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

Prerequisites

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Who Should Attend

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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.

Follow-On Courses

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Course Delivery

This course is available in the following formats:

Virtual Classroom Live

Experience live, expert-led online training from the convenience of your home, office or anywhere with an internet connection.

Duration: 4 day

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