Voice over IP Foundations: VoIP Training 5-day hands-on telecom training course, $2,895

Overview

This 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 training, participants 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 of training will focus on VoIP and IP telephony. The training is 60% practical 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 practical portion consists of 30 hands-on, skills-building labs where attendees will gain proficiency with some of the most popular VoIP software such as Wireshark, trixbox (formerly Asterisk@Home), Linksys Ethernet phone, SIP-based ATA, SIP-based Server and PBX products from Brekeke Software, Inc.

The training course includes 1-year access to a 50-book Online Networking Reference Library with titles specially selected to reinforce course concepts.

You'll Learn...

• Core concepts of how Internet Protocol (IP) carries a VoIP packet
• 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
• Security issues to consider when setting up VoIP

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

• Key architectural VoIP components
• End-to-end voice transmission
• Packetizing voice (encapsulation)
• Transmission time allocation
• QoS and capacity considerations
• Sources of delay
  • Coder processing delay (Think Time)
  • Algorithmic delay (Look Ahead)
  • Packetization delay
  • Serialization delay
  • Queuing delay
• Jitter buffer function
• VoIP QoS requirements: Packet Loss, Latency, Jitter

2. VoIP in the LAN

• MAC address
• IP address and ARP
• Ethernet switching
• Logical and physical segmentation
• VLAN - 802.1Q/P
• 802.3af - Power over Ethernet (POE)

3. IP Networking

• IP addressing
• Static routing
• OSPF
• EIGRP

4. TCP / IP and UDP

• Transmission Control Protocol (TCP)
• VoIP protocols that use TCP
• User Datagram Protocol (UDP)
• VoIP protocols that use UDP

5. SIP-Related IP Services

• DNS
• How SIP uses DNS
• DHCP
• How IP telephony uses DHCP

6. Voice Encoding and Compression

• G.711 u-law and A-law
• G.729
• G.723.1

7. Real-Time Transport Protocol (RTP)

• Dealing with Packet Loss, Latency, Jitter
• How various protocols defines the RTP session
  • Session Description Protocol
  • H.245 Terminal Capabilities
  • The RTP profile
  • The RTP payload type field
  • RTP telephony events (RFC 2833)
• How RTP removes jitter
• How RTP handles packet loss
• How RTP identifies the talking party
• How RTP handles silence suppression
• How RTP is used to mix voice (conference calls)
• The RTP header
• RTP Control Protocol (RTCP)
  • SDES
  • Sender/receiver reports
  • Bye reports

8. SIP Architecture

• SIP architecture
  • Proxy: stateful, stateless, call stateful, Session Border Controller
• SIP methods: INVITE, ACK, BYE, CANCEL, REGISTER, INFO, PRACK, etc.
• SIP response codes: 1xx, 2xx, 3xx, 4xx, 5xx, 6xx
• SIP headers (To:, From:, Call-ID:, Allows: Required, Via)
• Session Description Protocol (SDP)
• SIP Addressing, Session Control, and Call Setup

9. SIP Uniform Resource Indicators (URIs)

• Understand the format of SIP URIs and how URIs interoperate with PSTN dialing plans, email systems, and web pages
• Generic URI information (RFC 2396)
• Direct or Proxy
• PSTN number (RFC 2808)
• Instant messaging
• Presence
• In registrations

10. SIP Call Flow Examples

Review how SIP calls are set up for applications like PSTN, instant messaging, VoIP, and more in this technical, in-depth analysis of the protocol.

• Call attempt - unsuccessful
• Presence subscription
• Registration
• Presence notification
• Instant Message Exchange
• Call setup - successful
• Cancel
• Vacant number
• 100rel
• www authenticate

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
• 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)

• Architecture
• Verbs: CRCX, MDCX
• Responses
• Packages (DTMF, Line, Trunk, Generic, etc.)
• Parameter lines
• Sample call flow protocol analysis

15. H.323

• ASN.1 primer
• H.323 architecture
  • Gatekeeper
  • Gateway
  • MCU
  • Terminal
• H.323 versions
• H.323 gatekeeper controlled call flow example

16. Queuing

• Priority queuing
• Weighted fair queuing
• Weighted precedence
• Traffic policing and traffic shaping
• Low latency queuing
• The effects of data traffic and fair queuing on VoIP
• Mixing voice and data traffic effectively
• Determining bandwidth needs for voice traffic
• Assessing the impact of voice on data networks
  • Low speed links
  • High speed links

17. QoS Related Networking Protocols

• Differentiated Services (DiffServ)
• Call Admission Control

Labs (30)

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 a SIP Ethernet phone.

Lab 10: Configure the SIP Server.

Lab 11: Network SIP Proxies.

Lab 12: Configure a SIP softphone.

Lab 13: Configure a Wi-Fi radio.

Lab 14: Configure a Wi-Fi SIP phone.

Lab 15: Use Ethereal and Port Spanning to capture and analyze RTP.

Lab 16: Configure various CODECs and make test calls to compare voice quality (G.711, G.729, and G.723.1).

Lab 17: Reduce bandwidth consumption by 50% or more by increasing packet intervals and witness the QoS tradeoff.

Lab 18: CODEC bandwidth testing. Test the amount of bandwidth actually consumed by different types of voice compression.

Lab 19: Silence suppression and witness any QoS tradeoff. Activate and test silence suppression.

Lab 20: Configure automatic CODEC negotiation and observe how SIP negotiates CODECS (OFFER/ANSWER).

Lab 21: DTMF RFC 2833 and SIP INFO. Configure two different techniques that support accurate and reliable DTMF transmission.

Lab 22: Use Ethereal to capture and analyze RTCP (QoS) reports.

Lab 23: SIP REGISTER authentication. Configure a SIP phone to authenticate prior to joining a SIP network.

Lab 24: SIP INVITE authentication. Configure a SIP proxy to confirm the calling party prior to processing the call.

Lab 25: 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 26: Configure trixbox. Configure the system that promises " a PBX is 30 minutes". Configure SIP extensions, voice mail, and meetme conference.

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

Lab 29: Configure diff-serv on your VoIP gateway.

Lab 30: 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
The per student registration fee for this seminar is $2,895, and includes the seminar, course materials, and morning and afternoon refreshments.

Class 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 a week in advance of a class, please bring your confirmation letter. Business casual attire is appropriate.

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

Seminar Schedule

May 19-23	Raleigh, NC	Cary Training Ctr ()
May 26-30	Toronto, ON	Toronto Training Ctr ()
June 2-6	New York, NY	New York Offices ()
June 16-20	Washington, DC	Arlington Training Ctr ()
June 23-27	Morristown, NJ	Morristown Offices ()
July 7-11	San Jose, CA	Santa Clara Offices ()
July 14-18	Atlanta, GA	Atlanta Training Ctr ()
July 21-25	Sacramento, CA	E-Business ()
July 28-Aug 1	Boston, MA	Microtek ()
Aug 4-8	Dallas, TX	Irving training Ctr ()
Aug 4-8	Chicago, IL	Schaumburg Offices ()
Aug 11-15	Washington, DC	Arlington Training Ctr ()
Aug 18-22	San Jose, CA	Santa Clara Offices ()
Aug 18-22	Denver, CO	Interlocken Center ()
Aug 25-29	New York, NY	New York Offices ()
Aug 25-29	Columbus, OH	Dublin Training Ctr ()
Aug 25-29	Toronto, ON	Toronto Training Ctr ()

To learn about a similar seminar,
Click here to learn about , dates & locations

Payment is due prior to the 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, 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.

Contact Center Seminars
Telecom Seminars

Help Desk Books
Call Center Management
Tech Support Resources
Contact Center Technology
Online Support
Customer Satisfaction
Knock Your Socks Off
Help Desk Institute
Telecom Books
Communication Skills
Call Center Monitoring
Metrics & Benchmarking
CRM
Hiring, Training, & Retention
Outbound Telebusiness
Novelties, Gifts, & Humor
Bargain Books (50% off)

Subject Index
Catalog Index

Ways to Order
Shipping Options
About Us
Contact Us