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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.

  • Course Start Date: 2022-08-02
  • Time: 08:30:00 - 15:30:00
  • Duration: 4 days 08:30 AM - 03:30 PM
  • Location: Virtual
  • Delivery Method(s): Virtual Instructor Led
REGULAR PRICE $2,495.00 Save $249.50
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Course Outline


TCP / IP Networking


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 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
  • 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
             - 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

  • Lab 1: Network Hardware Installation
             - Install the network hardware.
  • Lab 2: Cisco IOS Command Line Interface Configuration
            - Configure Cisco IOS Command Line Interface via Telnet and console port access.
  • Lab 3: Configure VLAN
            - Configure VLAN for secure voice and data separation.
  • Lab 4: IP Network Configuration
             - Configure an IP network using static routing
  • Lab 5: Implement DNS
             - Configure a DNS zone, NAPTR, SRV, and A records as needed to support VoIP services.
  • Lab 6: Implement DHCP
             - Configure DHCP services on your LAN to support VoIP gateways and phones.
  • Lab 7: Calling without a SIP Proxy
             - Call without a SIP proxy
  • Lab 8: CORE Proxy Registration
             - Register with a CORE proxy
  • Lab 9: VoIP Island Configuration
             - Configure VoIP islands
  • Lab 10: SIP Ethernet Phone Configuration
             - Configure a SIP Ethernet phone
  • Lab 11: SIP Server Configuration
             Configure a SIP Server
  • Lab 12: Dial Plan Implementation
             - Implement the Dial Plan
  • Lab 13: SIP Softphone Configuration
             - Configure a SIP softphone
  • Lab 14: Capturing and Analyzing RTP using Wireshark
             - Use Wireshark and Port Spanning to capture and analyze RTP
  • Lab 15: Codec MOS Testing
             - Configure various codecs and make test calls to compare voice quality (G.711, G.729, and G.723.1)
  • Lab 16: Increasing Packet Intervals
             - Reduce bandwidth consumption by 50% or more by increasing packet intervals and witness the  QoS tradeoff
  • Lab 17: Codec Bandwidth Testing
             - Test the amount of bandwidth actually consumed by different types of voice compression
  • Lab 18: Silence Suppression
             - Silence suppression and witness any QoS tradeoff. Activate and test silence suppression
  • Lab 19: Codec Negotiation (Offer / Answer)
             - Configure automatic codec negotiation and observe how SIP negotiates codecs (OFFER / ANSWERS)
  • Lab 20: DTMF RFC 2833 and SIP INFO
             - Configure two different techniques that support accurate and reliable DTMF transmission
  • Lab 21: RTCP
             - Use Wireshark to capture and analyze RTCP (QoS) reports
  • Lab 22: SIP REGISTER Authentication
             - Configure a SIP phone to authenticate prior to joining a SIP network
  • Lab 23: SIP INVITE Authentication
            - Configure a SIP proxy to confirm the calling party prior to processing the call
  • Lab 24: SIP Call Flow Analysis
             - Using Wireshark, analyze typical call processing such as a normal call, busy call, abandoned call, and call transfer. Learn how to use Wireshark to troubleshoot problems with call processing
  • Lab 25: Wi-Fi Radio Configuration
             - Configure a Wi-Fi radio
  • Lab 26: Wi-Fi SIP Phone Configuration
             - Configure a Wi-Fi SIP phone
  • Lab 27: SIP Trunking
             - Configure SIP trunking between two SIP PBXs, and learn the process of connecting to the PSTN using ITSP rather than buying your own PSTN gateways and connecting using conventional TDM or analog methods
  • Lab 28: trixbox Meet-Me Conferencing
  • Lab 29: trixbox Voice Mail
  • Lab 30: SNMP SolarWinds Configuration
             - Install SolarWinds Engineer's Edition and use WAN Killer and SolarWinds SNMP tools to test QoS performance
  • Lab 31: VoIP Gateway DiffServ Configuration
             - Configure DiffServ on your VoIP gateway
  • Lab 32: Queuing Strategies and QoS Configuration
             - Configure various queuing strategies, apply service policies on your router, and witness the results. Perform file transfer and voice services on the same network and witness the results of proper and poor QoS configuration.

Assess Your Skills

TCP / IP Networking or equivalent knowledge is recommended before taking this course. Assess your skills with our free TCP / IP Networking Pre-Assessment Test.

Cancellation Policy

We require 16 calendar days notice to reschedule or cancel any registration. Failure to provide the required notification will result in 100% charge of the course. If a student does not attend a scheduled course without prior notification it will result in full forfeiture of the funds and no reschedule will be allowed. Within the required notification period, only student substitutions will be permitted. Reschedules are permitted at anytime with 16 or more calendar days notice. Enrollments must be rescheduled within six months of the cancel date or funds on account will be forfeited.

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About Global Knowledge


Global Knowledge is the world's leading learning services and professional development solutions provider. We deliver learning solutions to support customers as they adapt to key business transformations and technological advancements that drive the way that organizations around the world differentiate themselves and thrive. Our learning programs, whether designed for a global organization or an individual professional, help businesses close skills gaps and foster an environment of continuous talent development.

Training Provider Rating

This vendor has an overall average rating of 4.38 out of 5 based on 431 reviews.

I would never take another course that starts at 11AM and goes to 9PM again. The way the course was laid out really took away from ... Read more

I would never take another course that starts at 11AM and goes to 9PM again. The way the course was laid out really took away from the capturing of what was presented as it was 5-6 hours of watching a screen before getting to the actual labs. There has to be a better way to lay out this particular course. In my previous course, the lectures were broken up by labs which worked out fantastic and kept you engaged in the course. There were days when in order to actually complete the labs, would go over the 9PM day end time frame. Was able to get the primary labs done, but if you want to get all the content completed, you cannot complete it in the window of this course, you will need to come back on your own time.

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Instructor was great
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Wasn’t as advanced as I thought it would be. There was an issue when the day my course was the first time they used a new platfo ... Read more

Wasn’t as advanced as I thought it would be. There was an issue when the day my course was the first time they used a new platform.. from adobe to something called zoom; I had to call support line cause it stated our instructor wasn’t present. Thankfully I called cause everyone online was in the adobe virtual classroom waiting for what looked like a teacher who didn’t show up for class (IT didn’t get anything resolved until 10mins after start time). I felt like he was really getting hung up on very basic knowledge for the first half of the course (talking about how to create tabs and drag formulas as an example). I completed files a few times before he was done explaining. There was a scheduled fire drill for them (roughly 30mins)that also cut into our time, which wasn’t deducted from the hour lunch break or the two, fifteen min breaks. I also really wish he touched base more on the automating workbook functions portion which we barely did. I'm happy there were/are those study guides (learning videos) and exams to take on my own time that I hope after I've had the class are still available for me to learn from.

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