Voice over IP Foundations: VoIP Training
5-day training seminar, $2,995
5-day virtual (no travel) training course, $2395
RCCSP
  Professional
    Education
       Alliance

Overview

What You Will 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)
  • The H.323 protocol suite, including H.225, RAS, and H.245
  • 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

This recently updated hands-on VoIP training course provides essential VoIP and data networking knowledge including how VoIP works, why VoIP works, and how to use it. On the first day of Voice over IP training course, participants will configure an IP network using Cisco routers and switches, learning IP fundamentals that make VoIP easier to understand. The remaining four days of training will focus on VoIP and IP telephony.

This VoIP training is 60% hands-on labs and 40% lecture. The lecture portion of the training course uses technically detailed slides that illustrate the subject matter. Text-only slides are kept to a minimum. The hands-on portion consists of skill-building labs where attendees will gain proficiency with some of the most popular VoIP software products such as Wireshark, trixbox (formerly Asterisk@Home), Linksys Ethernet phone, SIP-based ATA, SIP-based Server and PBX products from Brekeke Software, Inc.

Who Should Participate

This training is is ideal 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.

Agenda

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
  • 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. 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

7. 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

8. 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

9. 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

10. 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

11. SIP Syntax

  • Request Message
  • Response Message
  • The Start Line
  • Via Header
  • SIP Dialog
  • From Header
  • To Header
  • Call-ID Header
  • Dialog State
  • CSeq Header
  • Max-Forwards Header
  • Proxy-Authenticate Header
  • Contact Header
  • Expires Header
  • User-Agent Header
  • Content-Length Header
  • Allow Header
  • Supported Header
  • Content-Type Header

12. 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

  • 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. H.323

  • System Architecture
    • H.323 Components
    • H.323: Umbrella Standard
    • Abstract Syntax Notation One
    • How H.225 Uses Q.931
  • H.323 Terminal Equipment
    • H.323 Gateway
  • H.323 MCU Components
  • MCU Controlled Three-Party Conference Example
    • MCU Controlled Transcoding
    • The Role of the Master
    • Terminal Type Identifiers for H.323 Master/Slave Determination
  • H.323 Protocols
  • Gatekeeper Control
    • Gatekeeper Discovery Multicast Method
    • Gatekeeper Multicast Filtering
    • GCF
    • RRQ
    • All Components Must Register if a GK is Utilized
    • Gatekeeper Direct Endpoint Routing Example
    • ADMISSION REQUEST
    • ADMISSION CONFIRM
  • SETUP
    • CALL PROCEEDING
    • Called Party ADMISSION REQUEST
    • Called Party ADMISSION CONFIRM
    • ALERTING
    • CONNECT
    • Direct Endpoint Signaling
    • Gatekeeper-Controlled Routing
  • Fast Start
    • Fast Start Procedure
  • H.323 and RTP
    • Gatekeeper Routed Trace (1-10)
    • Gatekeeper Routed Trace (11-19)
    • Gatekeeper Routed Trace (20-29)

16. Queuing

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

17. QoS Related Networking Protocols

  • 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

Dates, Locations and Registration

Prerequisites

TCP/IP Networking

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Hands-on Labs

1. Network Hardware Installation

  • Install the network hardware.

2. Cisco IOS Command Line Interface Configuration

  • Configure Cisco IOS Command Line Interface via Telnet and console port access.

3. Configure VLAN

  • Configure VLAN for secure voice and data separation.

4. IP Network Configuration

  • Configure an IP network using static routing.

5. Implement DNS

  • Configure a DNS zone, NAPTR, SRV, and A records as needed to support VoIP services.

6. Implement DHCP

  • Configure DHCP services on your LAN to support VoIP gateways and phones.

7. Calling Without a SIP Proxy

  • Call without a SIP proxy.

8. UA Registration

  • Register a UA with a proxy.

9. VoIP Island Configuration

  • Configure VoIP islands.

10. SIP Ethernet Phone Configuration

  • Configure a SIP Ethernet phone.

11. Networking SIP Proxies

  • Network SIP Proxies.

12. Dial Plan Implementation

  • Implement the Dial Plan.

13. SIP Softphone Configuration

  • Configure a SIP softphone.

14. Capturing and Analyzing RTP using Wireshark

  • Use Wireshark and Port Spanning to capture and analyze RTP.

15. Codec MOS Testing

  • Configure various codecs and make test calls to compare voice quality (G.711, G.729, and G.723.1).

16. Increasing Packet Intervals

  • Reduce bandwidth consumption by 50% or more by increasing packet intervals and witness the QoS tradeoff.

17. Codec Bandwidth Testing

  • Test the amount of bandwidth actually consumed by different types of voice compression.

18. Silence Suppression

  • Silence suppression and witness any QoS tradeoff. Activate and test silence suppression.

19. Codec Negotiation (Offer/Answer)

  • Configure automatic codec negotiation and observe how SIP negotiates codecs (OFFER/ANSWER).

20. DTMF RFC 2833 and SIP INFO

  • Configure two different techniques that support accurate and reliable DTMF transmission.

21. Using Wireshark for Capture and Analysis

  • Use Wireshark to capture and analyze RTCP (QoS) reports.

22. SIP REGISTER Authentication

  • Configure a SIP phone to authenticate prior to joining a SIP network.

23. SIP INVITE Authentication

  • Configure a SIP proxy to confirm the calling party prior to processing the call.

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.

25. Wi-Fi Radio Configuration

  • Configure a Wi-Fi radio.

26. Wi-Fi SIP Phone Configuration

  • Configure a Wi-Fi SIP phone.

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.

28. Trixbox Meet-Me Conferencing

29. Trixbox Voice Mail

30. QoS Performance Testing

  • Install SolarWinds Engineer's Edition and use WAN Killer and SolarWinds SNMP tools to test QoS performance.

31. VoIP Gateway DiffServ Configuration

  • Configure DiffServ on your VoIP gateway.

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.

Registration Fees

Both classroom training and virtual (online) training formats are available.

The per student registration fee for the hands-on classroom session is $2,995, and includes the seminar, course materials, and morning and afternoon refreshments.

The registration fee for virtual training session is $2,395.

Classoom training begins at 8:30 AM each day and concludes at 4:30 PM unless otherwise directed. Please arrive early on the first day to sign-in and meet fellow attendees. If you register less than one week in advance of a class, please bring your confirmation letter. Business casual attire is appropriate. For information on virtual training session start and stop times, refer to the course schedule below.

Register securely online with confidence or please call (708) 246-0320.

Seminar Schedule
Nov 17-21, 2014 Dallas, TX Irving Training Center
Dec 1-5, 2014 12:00-5:00 PM ET Virtual Classroom
Dec 8-12, 2014 Atlanta, GA Atlanta Offices
Dec 15-19, 2014 Washington, DC Arlington Offices
Jan 19-23, 2015 12:00-5:00 PM ET Virtual Classroom
Jan 26-30, 2015 Raleigh, NC Cary Training Ctr
Feb 2-6, 2015 Chicago, IL Schaumburg Offices
Feb 9-13, 2015 New York, NY New York Offices
Mar 2-6, 2015 12:00-5:00 PM ET Virtual Classroom
Mar 9-13, 2015 San Jose, CA Santa Clara Offices
Mar 23-27, 2015 Morristown, NJ Morristown Offices
Mar 30-Apr 3, 2015 Atlanta, GA Atlanta Offices

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Terms & Conditions

Seminar provider is not responsible for losses due to cancellation. In all circumstances, seminar provider's liability shall be limited to fees received.

Seminar agenda and assigned instructors are subject to change.

Public Training Terms & Conditions

Payment is due prior to the seminar.

Public seminar cancellation policy.  Registrants may cancel up to fourteen days in advance of the seminar start date for a full refund, less administrative fees of $500.  Or, you may transfer your registration to another member of your company at no additional charge.  Registrants canceling within fourteen days of the seminar will receive training credit, less administrative fees of $500 toward any other Resource Center seminar.

In the unlikely event that a seminar must be cancelled by seminar provider due to unavoidable circumstances, you will be notified at least two weeks prior to the seminar date, and your payment will be refunded.  Seminar provider is not responsible for losses due to cancellation including losses on advanced purchase airfares.  We strongly recommend that attendees traveling by air to attend the seminar purchase only refundable tickets.Become a certified callcenter manager