Voice over IP Foundations 5-day hands-on telecom training course, $2995 5-day virtual (no travel) training course, $2395

Overview

This recently updated hands-on 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 training, 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.

The training is 60% hands-on labs and 40% lecture. The lecture portion of the training 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.

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)
• 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

Who Would Benefit
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

• Teleph3ony 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

Hands-on Labs

Lab 1: Install the network hardware.

Lab 2: Configure Cisco IOS Command Line Interface via Telnet and console port access.

Lab 3: Configure VLAN for secure voice and data separation.

Lab 4: Configure an IP network using static routing.

Lab 5: Configure a DNS zone, NAPTR, SRV, and A records as needed to support VoIP services.

Lab 6: Configure DHCP services on your LAN to support VoIP gateways and phones.

Lab 7: Call without a SIP proxy.

Lab 8: Register a UA with a proxy.

Lab 9: Configure VoIP islands.

Lab 10: Configure a SIP Ethernet phone.

Lab 11: Network SIP Proxies.

Lab 12: Implement the Dial Plan.

Lab 13: Configure a SIP softphone.

Lab 14: Use Wireshark and Port Spanning to capture and analyze RTP.

Lab 15: Configure various codecs and make test calls to compare voice quality (G.711, G.729, and G.723.1).

Lab 16: Reduce bandwidth consumption by 50% or more by increasing packet intervals and witness the QoS tradeoff.

Lab 17: Test the amount of bandwidth actually consumed by different types of voice compression.

Lab 18: Silence suppression and witness any QoS tradeoff. Activate and test silence suppression.

Lab 19: Configure automatic codec negotiation and observe how SIP negotiates codecs (OFFER/ANSWER).

Lab 20: Configure two different techniques that support accurate and reliable DTMF transmission.

Lab 21: Use Wireshark to capture and analyze RTCP (QoS) reports.

Lab 22: Configure a SIP phone to authenticate prior to joining a SIP network.

Lab 23: 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: Configure a Wi-Fi radio.

Lab 26: Configure a Wi-Fi SIP phone.

Lab 27: 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: Install SolarWinds Engineer's Edition and use WAN Killer and SolarWinds SNMP tools to test QoS performance.

Lab 31: Configure DiffServ on your VoIP gateway.

Lab 32: 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

Feb 15-19, '10	12:00-5:00 PM ET	Virtual Classroom
Feb 15-19, '10	Washington, DC	Arlington Training Ctr ()
Feb 22-26, '10	Boston, MA	Microtek ()
Feb 22-26, '10	Toronto, ON	Toronto Training Ctr ()
Mar 1-5, '10	Dallas, TX	Irving training Ctr ()
Mar 8-12, '10	Orlando, FL	Microtek ()
Mar 15-19, '10	San Jose, CA	Santa Clara Offices ()
Mar 22-26, '10	Chicago, IL	Schaumburg Offices ()
Mar 29-Apr 2, '10	San Francisco, CA	Microtek ()
April 5-9, '10	Morristown, NJ	Morristown Offices ()
Apr 12-16, '10	Washington, DC	Arlington Training Ctr ()
Apr 12-16, '10	Los Angeles, CA	Microtek ()
Apr 19-23, '10	12:00-5:00 PM ET	Virtual Classroom
Apr 19-23, '10	New York, NY	New York Offices ()
Apr 26-30, '10	Raleigh, NC	Cary Center ()
May 3-7, '10	Philadelphia, PA	Microtek ()
May 10-14, '10	Atlanta, GA	Atlanta Training Ctr ()
May 17-21, '10	Toronto, ON	Toronto Training Ctr ()
May 17-21, '10	Dallas, TX	Irving Training Ctr ()
May 24-28, '10	Houston, TX	Texas Conference Ctr ()
May 24-28, '10	San Jose, CA	Santa Clara Offices ()
Jun 14-18, '10	Chicago, IL	Schaumburg Offices ()
Jun 21-25, '10	12:00-5:00 PM ET	Virtual Classroom
Jun 21-25, '10	Washington, DC	Arlington Training Ctr ()
Jun 28-Jul 2, '10	Boston, MA	Microtek ()
Aug 23-27, '10	12:00-5:00 PM ET	Virtual Classroom
Oct 25-29, '10	12:00-5:00 PM ET	Virtual Classroom
Dec 27-31, '10	12:00-5:00 PM ET	Virtual Classroom

, dates & locations.

Payment is due prior to the seminar.  If payment is not received, a credit card hold will be required for participation.  This card will only be processed if payment has not been received within two weeks following the seminar.

Cancellation Policy.  Registrants may cancel up to fourteen days in advance of the seminar start date for a full refund.  Cancellations within fourteen days of the seminar start date will be subject to an administrative fee of $500.

In the unlikely event that a seminar must be cancelled, you will be notified at least two weeks prior to the seminar date.  Seminar provider is not responsible for losses due to cancellation including losses on advanced purchase airfares.  As seminars are cancelled for under-enrollment from time to time, we strongly recommend that registrants traveling by air purchase only refundable tickets.

Seminar agenda subject to change. Errors and omissions in pricing are not accepted.

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