Implementing Voice over IP
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Overview

Voice over IP is rapidly moving from a tactical cost saving effort to a more long term strategy of productivity improvements and reduced cost of network ownership. The ability to service voice communications needs over your existing data networks is now a reality. You can maximize your savings through the critical elements of proper evaluation and design.

This course provides real-world, multi-vendor options for integrating voice and data communication applications. You will analyze cost versus. call quality issues and understand the key standards and technologies that make VoIP a reality.

In our intensive hands-on labs, you will evaluate public internet calling, bandwidth considerations, echo control, jitter, voice compression, softswitch function and more. Whether you are considering fully deploying VoIP or implementing IP Telephony in a hybrid approach, this course will help you understand thet options you face, problems you will encounter, and the security and performance issues you'll need to consider.

You'll Learn:

  • Core concepts of how IP (Internet Protocol) carries a VoIP packet
  • The benefits and capabilities of SIP (Session Initiation Protocol)
  • How to implement RTP and QoS to ensure the highest voice quality over your IP networks
  • The essentials of signaling and when to use SIP, MEGACO, H.323, or MGCP
  • Understand how SIP establishes, modifies and terminates "sessions" over IP networks
  • The importance of QoS (jitter, packet delay, packet loss)
  • Compare IP, ATM, and Frame Relay voice/data networks
  • Decipher the call setup procedure under the H.323 standard
  • Protocol flows and how to analyze code negotiation using network trace tools
  • How a VoIP gatekeeper acts as a virtual telephony switch
  • Security issues to consider when setting up your VoIP
  • The missing pieces of VoIP implementation, including signaling, call accounting, and billing
  • The effect of line jitter, call latency, compression, and sockets
  • Evaluate PC-to-PC, PC-to-phone, and phone-to-phone calls

Who needs to attend

IT managers, technical sales/marketing personnel, consultants, network designers, network engineers, product design engineers developing integrated-services products, telecom technicians and managers needing to understand Voice over IP, and systems administrators who will manage a converged network.

Look at this agenda!

Packetizing Voice

  • Transmission, Switching, and Signaling
    • Introduction to the VoIP Standards
    • Time Division
    • G.711 Pulse Code Modulation
    • Compare u-law and A-law
  • Transmission in VoIP networks
    • The 20 millisecond Voice Packet
    • The 60 ms Voice Packet
    • The Voice Packet Header
    • Voice Packet Analysis
    • Voice Packet Analysis: Other Voice Packet Sample Sizes
  • Introduction to Packet Switching
    • The ISO OSI Reference Model
    • Addressing
    • Sample RTP Packet
    • Packet Switching
  • Voice Compression
    • CODECS
  • Linear Predictive Coding
    • G.729 CS-ACELP
    • Calculating Bandwidth: Calculate Overhead First
    • Calculating Bandwidth: Calculate Payload Next
    • Payload and Overhead Equals Total Bandwidth

TCP/IP Review

  • Transmission Control Protocol vs. User Datagram Protocol
  • Connection-Oriented Protocol (Transmission Control Protocol)
  • TCP Slow Start Congestion Window Size Behavior
  • TCP/IP Packet Format and Operation
  • Connectionless Protocols (User Datagram Protocol)
  • UDP Packet
  • DHCP (Dynamic Host Configuration Protocol)
  • DNS
    • Basic Method of DNS
    • Resource Records
  • Multicast
    • Multicast Support for Emerging VoIP Applications
    • Unicast, Broadcast, and Multicast Addresses Compared
    • Multicast Looks Like Broadcast on Layer 2
    • Multicasting with and without IGMP Snooping
  • The traceroute Command
    • Internet Control Message Protocol Technique
  • LDAP

Real-Time Transport Protocol

  • Real-Time Transport Protocol Architecture
    • Real-Time Transport Protocol and RTP Control Protocol
    • RTP Ports
    • Some RTP Uses
  • RTP Profile
    • Payload Types
    • NTP vs. RTP Timestamp
    • RTP Timestamps
    • RTP Timestamps and Silence Suppression
    • RTP Timestamps and Jitter Calculation
  • Mixers
    • Synchronization Source
    • Mixers Add Contributing Source Fields
  • RTP Header
    • UDP Packet with RTP Header and Voice
    • Required Fields
    • Version
    • Padding Bit
    • Extension Bit
    • CSRC
    • Marker Bit
    • Payload Type
    • Sequence Number
    • Timestamp
    • SSRC
  • RTP Related Standards
    • SSRC
    • Call Stacking: The Key to Efficiency
    • RTP Related Standards
  • RTCP
    • RTCP QoS: Round Trip Delay Calculation
    • Sender Reports
    • Receiver Reports
    • Source Descriptions
    • Source Description Items
    • Other RTCP Packets

Technology Necessary to Make VoIP Successful

  • Controlling Delay
  • Sources of Delay
    • Coder Processing Delay (Think Time)
  • Algorithmic Delay (Look Ahead)
    • Packetization Delay
    • Serialization Delay
    • Queuing Delay
    • Network Switching Delay: MIQ Ratings
  • Controlling Jitter
    • Jitter Buffer Delay
    • Latency Example without LFI
    • Multilink PPP with LFI
  • LFI Function
    • Interleaving on a Gigabit Ethernet Link
    • Latency Example with LFI
    • LFI Matrix
  • Controlling Serialization Delay
    • Perfect Candidate for LFI and RTP Header Compression
  • RTP Header Compression Process
    • RTP Header Compression Format
  • Controlling Queuing Delay
    • Low Latency Queuing
    • Connection Admission Control
    • RSVP East Path and West Reserve Messages
    • RSVP is Half Duplex
    • RSVP West Path and East Reserve Message
    • RSVP Reservation Established
    • RSVP Synchronization QoS before the Phone Rings
    • Differentiated Services RFC 2474
    • IP Type of Service Field
    • IP ToS Field Becomes the DSCP
    • Inline Power
  • Virtual Private Networks
    • Message Privacy
    • VPN Tunnel
    • Echo Control

H.323

  • System Architecture
    • H.323 Components
    • H.323 Umbrella Standard
  • H.323 Terminal Equipment
    • H.323 Gateway
    • MCU Controlled Transcoding
    • Terminal Type Identifiers for H.323 Master/Slave Determination
  • H.323 Protocols
    • H.323 Voice over Internet Protocol Packet Trace
  • 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
    • Gatekeeper Controlled Routing Example
    • Gatekeeper Routing across Multiple Zones
  • Fast Start
    • Fast Start Procedure
  • Qos and H.323
    • RSVP Synchronization QoS before the Phone Rings

Session Initiation Protocol

  • SIP Architecture
    • IETF Standards
    • SIP vs. H.323
    • SIP and Other Protocols
    • SIP Naming Standard
    • SIP Methods (Requests)
    • SIP Response Codes
    • SIP Uniform Resource Indicators
    • SIP Components
    • SIP Registration
  • SIP Protocol
    • SIP Proxy Server Example
    • SIP INVITE
    • Via Field
    • Via Operation
    • Loop and Spiral Detection
    • Routing of Subsequent Requests
    • Max-Forwards
    • SIP Dialog (Formerly Call Leg)
    • Command Sequence
    • Contact
    • Content
    • SDP: RFC 2327
    • Negotiating the Session
    • 180 Ringing
    • 200 OK
  • SIP Reliability
    • SIP Reliability on Invite
    • SIP Reliability on BYE
  • Stateful vs. Stateless
    • Hop-by-Hop vs. End-to-End
    • Stateful vs. Stateless Proxies
  • SIP Mobility
    • SIP Redirect Server
    • SIP Forking
    • CANCEL
  • SIP Products
    • Cisco SIP IP Phone: Model CP7960
    • Pingtel Xpressa SIP Phone
  • RTSP
    • RTSP: RFC 2326
    • RTSP Decode Sample
    • Audio/Video Mail Integration with RTSP, SIP, and E-mail for Call Centers

Gateway Protocols

  • Introduction to Gateway Control Protocols
    • Session Initiation Protocol vs. Softswitch (MGCP and MEGACO)
    • MEGACO and H.248
    • Total Decomposition of the CO Switch
    • Move All Dial-Peers to the MGC
  • Why MEGACO?
    • MEGACO Cost Savings
    • MEGACO: Why Not?
    • MEGACO: A Transitional Strategy
  • MEGACO Architecture
    • MEGACO Access Gateway Architecture
    • Traditional TDM Trunking
  • MEGACO Trunk Gateway Architecture
    • MEGACO Connection Model
  • MEGACO Protocol
    • Packages Equal Interoperability
    • Package Examples
    • DigitMaps
    • MEGACO Commands
    • MEGACO Transactions
    • MEGACO Protocol Example
  • Gateway Protocol History
    • Internet Protocol Device Control
    • Simple Gateway Control Protocol
    • MGCP
    • Media Device Control Protocol
    • H.GCP
    • MEGACO
    • H.248
    • Agreement between MEGAGO and ITU SG16

Network Security

  • Firewall Architecture
    • What Is a Firewall?
    • The Weakness of Firewalls
    • The Demilitarized Zone
    • Categories of Firewalls
    • Packet Filter
    • SYN Attack: Denial of Service
    • SYN Attack: Stateful Firewall Protection
    • The Architecture of Proxy Servers
  • NAT
  • NAT Fails on Real-Time Transport Protocol
  • Firewall Vendors
    • Commercial Firewalls

Implementing VoIP

  • Cisco AVVID: Architecture for Voice, Video, and Integrated Data
  • Avaya MultiVantage
    • Avaya Retrofits the G3R
  • Nortel Networks Succession CSE 1000
    • Nortel Networks ITG and Succession Media Services Cards
    • Nortel Networks Business Communications Manager

Course Labs:

Lab 1: Install the Network Hardware

Lab 2: Cisco IOS Command Line Interface

Lab 3: Configure an IP Network

Lab 4: Using the TFTP Server

Lab 5: POTS dial-peers

Lab 6: VoIP dial-peers

Lab 7: Solving a One-way Voice Problem

Lab 8: Show Call Active Voice

Lab 9: Make a Public Internet Call

Lab 10: CODEC MOS Test

Lab 11: VAD Lab

Lab 12: CODEC Pool lab

Lab 13: Phone to PC lab

Lab 14: PC to Phone lab

Lab 15: PC to PC w/video

Lab 16: Echo Control lab

Lab 17: Adaptive Jitter Buffer

Lab 18: Packet Interval

Lab 19: Bandwidth requirements

Lab 20: RTP header compression

Lab 21: VoIP Bind Command

Lab 22: Navigate the BCM

Lab 23: Ethernet phone implementation

Lab 24: QoS with RSVP

Lab 25: SIP

Lab 26: Gatekeeper Controlled dial-peers

Lab 27: H.323 Trace

Lab 28: MEGACO Trace

Special bonuses for on-line registration!

Register from this web site and receive a complimentary telephony book from the Resource Center. Choose from:  

Click here to learn more The Telephony Tutorials
Click here to learn more Telecommunications Projects Made Easy
McGraw-Hill Illustrated Telecom Dictionary
Business Telecom Systems
Click for more information! Microsoft Internet & Networking Dictionary
Click here to learn more! Network Tutorial

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Registration Fees
The per student registration fee for this seminar is $1,995, and includes the seminar, course materials, and morning and afternoon refreshments.

To register, click on the "Book Now" button or please call (708) 246-0320

Seminar Schedule
This Class is no longer available.

The Training Center opens every day at 7:30 AM. You must sign-in with the receptionist on the first day of class. If you register less than a week in advance of a class, please bring your confirmation letter. Classes begin at 8:30 AM each day and conclude at 4:30 PM unless otherwise directed. Business casual attire is appropriate.


Payment is due prior to the conference.

Cancellation Policy.  Due to the preparations required for this multiple-lab seminar, registrants are expected to attend the seminar at the location and date selected. If you can not attend, you may transfer your registration to another person at no additional charge and without penalties. Registrants may cancel up to forty-five days in advance of the seminar start date for a full refund, less administrative fees of $400.  There will be no refunds or credits for cancellations made within forty-five days of the seminar or for non-attendance. Please be sure you can attend before registering.

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.


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PO Box 401, Western Springs, IL  60558
Tel: (708) 246-0320   Fax: (708) 246-0251  

Copyright © 2003-2006  Resource Center for Customer Service Professionals.  All rights reserved.
Last modified March 31, 2006