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What can convergence do for your business ? Through converged voice and data networks, your business will find savings in network administration, bandwidth, performance, time and cost of moves, adds and changes.

In order for convergence to work, it is critical that you understand voice and data convergence components, how voice is sent through a packet-based network, circuit- and packet-switched networks, the real-time requirements of telephony, the reliability of PBX systems, and the implementations that make business communications more effective and productive.

In this 2-day course, you will learn how to integrate legacy phone systems with VoIP gateways, understand how voice is effectively carried in packets rather than circuits, and determine if your LAN and/or WAN will carry Voice over IP calls.

What You'll Learn

• Learn how SIP controls VoIP calls
• Learn how to provision a SIP proxy and SIP User Agents including SIP gateways, SIP authentication, and SIP softphones
• The legacy telephone signaling protocols, LS, GS, E&M, DID, and ISDN
• Learn about QoS technologies and how they ensure high quality voice
• How IP Networking relates to successful VoIP implementations
• Major QoS issues and QoS technology including DiffServ, RSVP, and 802.1q/p
• Learn QoS issues relative to today's WAN technologies
• Understand voice compression
• Hear and compare G.711, G.723, G.726, and G.729
• VoIP over Frame Relay, DSL, ATM, SONET, Cable Modem, and the Public Internet
• Calculate bandwidth needed to support VoIP traffic using Erlang-B analysis
• Learn the function of a softswitch in MGCP and MEGACO networks
• Learn how VoIP can function in SS7-controlled carrier networks
• The transmission of converged traffic over analog, digital, and optical transports
• To identify the speech compression standards under H.245, including G.723, G.726, and G.728
• To conduct traffic engineering and congestion management case studies using Erlang B tables

Who Would Benefit

This course is intended for 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.

Look at this agenda!

Voice Packetization

• General voice/data convergence architecture
• SIP architecture
• Circuit switching vs packet switching
• G.711 Pulse Code Modulation
• Voice packets
• Time division multiplexing versus packet division multiplexing

Voice Compression

• ASICs
• Mean Opinion Score (MOS)
• Listen to popular CODECs and rate them
• Adaptive Differential Pulse Code Modulation (ADPCM) method
• Linear predictive coding algorithm
• Demonstration of CODEC behavior under packet loss conditions

WAN Technology as it relates to Voice over IP

• SONET architecture
• SONET and ATM interworking
• Creation of virtual channels
• Definition of the ATM Permanent Virtual Circuit
• ATM VPI and VCI
• Connection Admission Control (CAC)
• Frame Relay and ATM interworking
• Access
• Port speed
• PVC (DLCI)
• CIR

Quality of Service (QoS)

• Technology options for QoS
• Leaky bucket
• Resource Reservation Protocol (RSVP)
• DiffServ
• Common Open Policy Service (COPS)
• 802.1q/p
• Multiprotocol Label Switching (MPLS)

Access Signaling Types

• Lines vs. trunks
• Trunk routes
• Loop start
• Ground start
• Direct Inward Dialing (DID) trunk
• E&M tie trunk
• T1 carrier
  • D4 Super Frame
  • B8ZS
  • D5 Extended Super Frame
• ISDN Q.931 signaling protocol
• How PRI and BRI use Q.931 signaling
• Q.931 call setup process
• Comparison of signaling protocols based on ISDN's Q.931
  • MEGACO, ATM Q.2931, H.323, SS7

Traffic Engineering

• Measuring engineering
• Grade of service
• Voice vs. data congestion
• Traffic model
• Erlang B
• Trunk efficiency
• Voice and data over shared bandwidth
• Measurement calculations

Course Labs

Lab 1: Install the Lab Network

You will interconnect your equipment with the rest of the classroom to create an IP network to begin testing VoIP protocols

Lab 2: Configure your PC

Configure your PC for IP networking, then test the network

Lab 3: DEMO Provision Router / DHCP Server / SIP Server

The instructor will demonstrate the provisioning of essential centralized services for DHCP, SIP, and basic routing

Lab 4: Provision SIP Media Gateways

Configure SIP gateways to illustrate typical requirements for a voice gateway. Determine your SIP gateway's IP address. Provision a SIP gateway's phone number and provision your SIP gateway's host name

Lab 5: Dialing without a SIP Proxy

Configure your SIP gateway to permit SIP URI dialing without the use of a SIP server. Observe the organizational value of assigning host names, and see how a VoIP network would operate without central control

Lab 6: Provision SIP Proxy and Digit Maps

You will provision your SIP gateway to use a SIP proxy, make calls using a SIP proxy, understand the value of a SIP proxy, and implement a single candidate digit map

Lab 7: Provision a SIP Softphone

Provision a SIP softphone on your PC and observe the similarities between a SIP softphone and a SIP gateway

Lab 8: CODEC Testing

You will provision several popular voice compression CODECS on your SIP gateways and then make test calls to hear the popular voice compression standards and judge the voice quality

Lab 9: SIP Security - Authentication

You will provision authentication on the SIP proxy and SIP gateway and explain the value of restricting registration to only authenticated SIP phones

Lab 10: SIP Security - Outbound Proxy

Force all outbound calls to be processed through the proxy and understand why using an outbound proxy can control calling activity and enforce another layer of security

Lab 11: QoS - DiffServ

After you configure your SIP gateway with the proper DiffServ codepoints, you will use the Ethereal analyzer to check your configuration.

Special bonuses for on-line registration!

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

The Telephony Tutorials
Telecommunications Projects Made Easy
McGraw-Hill Illustrated Telecom Dictionary
Business Telecom Systems
Microsoft Internet & Networking Dictionary
Network Tutorial

* Normal shipping and handling fees apply

Registration Fees

The per student registration fee for this seminar is $1,495, 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.