TECHtionary University

For access to 3,000+ animated tutorials, click here: http://techtionary.com

SIP 2012 Business Applications Planning Guide

SIP-Session Initiation Protocol

& Optional Introduction to Lync Communications Server

2-5 Days Onsite, Via Webseminar or ~32 hours online.  

NEW for 2012 – Designing and Planning Scalable SIP Networks

© TECHtionary Corporation – all rights reserved

For more information and scheduling, please call Tom Cross 303-594-1694

or cross@gocross.com

 

Volume discounts are available.  All major credit cards are accepted.  Special SIP Forum Member discounts are available.  Additional discounts for multiple sessions are available. For more information on SIP Forum go to www.sipforum.com

Reference Promotional Code – SIPF8203

What Providers, Agents and Users are saying about these courses:  

According to Matt Jolly IT Consultant, “SIP Business Executive course training to a new higher level. For example, there is nothing like the tutorials SIP available anywhere or from anyone. For the channel partner or customer, this course provides critical insights for successful implementation and management. The new user interface speeds learning allowing viewers to grasp complex concepts faster than ever before. With this course, SIP/VoIP providers can rapidly accelerate the learning process for their channel partners which in turn accelerate revenues. Now is the time for users and providers alike to make this course an integral part of their business operations.”

Matt Jolly – Consultant
……………………………………..
“I am writing regarding Tom Cross and the online training and VoIP/SIP course. By far this is the best program in place, as Tom is one of the most recognized trainers in the United States for VoIP/SIP. It might be worth to add this course to your sign-up package for agents around the country not only as a profit center but something that you could co-brand and have a “leg-up” on your competition.”

Bill Bowyer – CEO – VoIP in America

……………………………………
“The SIP/VoIP courses are more than a superb primer on VoIP/SIP technology; they are an indepth business plan for a VoIP/SIP implementation. In addition, the VoIP/SIP diagnostic and troubleshooting guide is the most thorough approach to VoIP QoS available. I need information that informs but does not overwhelm. Information that guides but not drives you away. The courses provide insights and actionable information that has helped me get the technology we need sooner but saved me a considerable amount in understanding what we didn’t need. The SIP course especially is a valuable one which provides much needed information in a readily understandable format.”

Paul Daubitz – President – ATI-TeleManagement

Who Should Attend:  – This online course is designed for enterprise executive and technical managers, channel partners, VAR-Value-Added Resellers, SI-Systems Integrators, telephone interconnect, agents, master agents and consultants.  In addition, this course will benefit corporate technical, staff marketing, training business development, sales, channel managers, operations, engineering, support and other corporate managers for SIP-VoIP providers, carriers, software developers and hardware manufacturers. 

What You Will Learn:

• Review the fundamentals of IP-Internet Protocol and platforms required for highperformance SIP-VoIP systems.  This includes soft switches, gateways, routers, services and other critical components.
• Explore business applications and opportunities.  Review what customers are buying today and why they are buying.  In addition, emerging “killer applications” will be explained in depth.
• Quickly grasp complex subjects such as H.323, MGCP and SIP.  As SIP-Session Initiation Protocol emerges are the key VoIP communications protocol, discover how this technology will impact all voice communications systems from key, PBX, IP-PBX, hosted, managed and other systems.
• Understand basic and advanced SIP-VoIP concepts features.  From hosted, managed, IAS, and IP-PBX, quickly understands “what’s-what” for different customer applications and business models.
•  Probe the issues behind Integrated and Converged Access.  Understand when and why organizations need a converged access solution.
• Understand why “network assessment” is critical to any SIP-VoIP implementation and why this step cannot be overlooked.
•  Address the issue of QoS-Quality of Service by overcoming jitter, echo, noise and other network problems.  Review the role of RTCP and other tools to monitor and maintain high performance VoIP networks.
• Understand the functions of the new communications “toolbar.”  See how the benefits of “unified communications” as they improve business operations.
• Assess the Top-10 issues why SIP trunking and hosted VoIP is more than “dial-tone,” and how it can represent change in the business and business model of even the smallest enterprises.  Discuss and explore new ways to improve fundamental business processes.
• Explore how a SIP-VoIP call is processed and review potential security attacks.   Discover how SPIT, VOMIT, Calljacking, DOS and other terrorist attacks can target not just data, but voice packets.
• Review SIP and SIP Trunking and all the implications and applications from TCO-Total Cost of Ownership to QoS-Quality of Service.  SIP Trunking is the most profound new form of telecommunications since POTS.
•  Explore Microsoft’s Lync Communications Server features, concepts, call flows, configurations and other issues for evaluation and implementation.

 

This ebook is provided to all students of the SIP 2012 Business Applications Planning Guide Course and some concepts from the book are included in the courseware. However, since there is more than four years of vendor-neutral continuous research and development, the SIP 2012 Course provides a more comprehensive view of the subject.

SIP Trunking is one of the first complete books to planning, evaluating, and implementing high-value SIP trunking solutions.  Most large enterprises have switched to IP telephony, and service provider backbone networks have largely converted to VoIP transport. But there’s a key missing link: most businesses still connect to their service providers via old-fashioned, inflexible TDM-Time Division Multiplexed trunks. Cisco^® authors show how to use Session Initiation Protocol (SIP) trunking to eliminate legacy interconnects and gain the full benefits of end-to-end VoIP.  Written for enterprise decision-makers, network architects, consultants, and service providers, this book demystifies SIP trunking technology and trends and brings unprecedented clarity to the transition from TDM to SIP interconnects. The authors “separate the true benefits of SIP trunking from the myths and help you systematically evaluate and compare service provider offerings.  This book includes detailed cost analyses, including guidance on identifying realistic, achievable savings.”  SIP Trunking also introduces essential techniques for optimizing network design and security, introduces proven best practices for implementation, and shows how to apply them through a start-to-finish case study.  Some of the concepts explored in the book are

• Discover the advanced Unified      Communications solutions that SIP trunking facilitates
• Systematically plan and prepare your      network for SIP trunking
• Generate effective RFPs for SIP      trunking
• Compare SIP deployment models and      assess their tradeoffs
• Address key network design issues,      including security, call admission control, and call flows
• Manage SIP/TDM interworking throughout      the transition

• This IP communications book is part of the Cisco Press® Networking Technology Series. IP communications titles from Cisco Press – SIP Trunking book by Christina Hattingh, Darryl Sladden, ATM Zakaria Swapan was first published February 4 – 2010 available at www.informit.com – ISBN-10: 1-58705-946-0

Quick Overview

Day 1 is optional for students who are less familiar with IP and VoIP networking.

In other words, look at the outline for the SIP course and if the students are not knowledgeable of the key concepts, then Day 1 is recommended.

Day 2 & 3 is the indepth SIP course. 

Day 4 is optional Lync intro course.

NOTE: All content can be re-organized or customized to suit specific learning needs.  Call to discuss.

Day One – Fundamentals of Data/Internet Telecommunications

NOTE: Day one is recommended for those wanting an update in internet technologies, data networking and network services (bandwidth).

-  Fundamental Network and IP Technologies – the IP in SIP

• 1 – Voice-to-Digital-to-Packet Transmission
• 2 – Back To Basics – Cabling, Conduit and Electrical Systems
• 3 – Transmission Concepts – DSL, T-1/E-1, ISDN-PRI, SIP Trunking, GIG-E
• 4 – Optical Fiber & Bandwidth
• 5 – Integrated Access Services – Dynamic Bandwidth Allocation – BOD-Bandwidth On Demand
• 6 – Introduction to IP-Internet Protocol and VoIP-SIP , MPLS-Multi-Protocol Label Switching, DiffServ-Differentiated Services, DSCP Differentiated Services Code Points and Packet Priority Classifications, TOS-Type of Service, EF-Express Forwarding, MPLS Uniform mode, MPLS Pipe and Short-Pipe modes, WRR-Weighted Round Robin, TCB-Traffic Conditioning Blocks – Marker, Meter, Shaping, Droppers and PHB-Per Hob Behavior.
• 7 – TCP/IP and other Protocols and Layers – RTP, RTCP, SDP, SOAP, SALT

-  Call processing with Route, Image, DHCP, DNS, Image, Configuration servers

• 8 – Hardware – Routers, Switches – MAC-Media Access Control, WiFi-VLANS-VPNS
• 9 – Protocols “Rules of the Road” – H.228, H.323, MGCP, SIP, and Desktop “Softphones,” “Toolbars” and other end points (desksets)
• 10 – IP-PBX and Hosted VoIP/SIP – Integrated/Unified/Homogenized

- Top-10 Critical Technologies to SIP

1 –  IP protocol, IP networking and a VPN

2 – The difference between IAS-Integrated Access Service versus Converged Access Service

- Enhanced IAS with MPPP-Multi-link Point-to-Point Protocol, PPP Multilink Protocol (MP), L2TP-Layer 2 Tunneling Protocol

- VPLS-Virtual Private LAN Service – new name for metro-Ethernet

- Switching Versus Routing – key benefits of VPLS

3 –  SIP-Session Initiation Protocol Trunking -

-        SIGTRAN (Signaling Transport)

-        SCTP-Stream Control Transmission Protocol

4 – Justification for an IP PBX – options and approaches

5 – Technical difference between IAS-Integrated Access Service, Hosted and Managed VoIP

-  Call processing with Route, Image, DHCP, DNS, Configuration servers

-  Media Gateways replace PBXs -  the following tutorials are some examples of customer applications of MG-Media Gateways:

- Connection of IP-PBX to PSTN
- Connection of IP-PBX to PSTN & SIP trunk provider
- Survivable connection to SIP trunk provider
- Connection of PBX & IP-PBX to PSTN & SIP trunk provider
- Connection of IP-PBX to Hosted VoIP provider
- Connection of IP-PBX & PSTN to Microsoft Lync Server

-  SC-Session Controllers or SBC-Session Border Controllers are access devices operate at Layer 5 Session Layer, whereas routers operate at Layer 3 Network.   Some of the key SBC/SC functions are:

- Secure network peering – private and public to enhance performance

- Topology hiding – using various types of inter-AS-Autonomous System features as well as separating media (voice) and hide signaling (IP addresses) and data streams (traffic)

- Border call routing – routing at AS level rather than with interior protocols

- Interoperability – access/restrict to reduce voice spam

- QoS & Call Admission Control – load/jitter correction

- Billing systems interoperability – reduce billing errors

- NAT-Network Address Translation – routing for maximum performance

- CALEA-Communications Assistance for Law Enforcement Act

- Compatibility with billing

- Dialect conversion

- Protocol conversion

- Codec conversion

- Firewall restrictions

- Wholesale and Transit peering

6 – “Open Source” PBX options

7 – QoS-Quality of Service importance – how to measure it and fix it

8 –  Softphones – Where they make sense – user benefits

9 -   The difference between IPT-Internet Protocol Telephony and VoIP – Cisco, Broadsoft, Lync and other platforms

10 -  Unified Communications – Mobility Applications

Day Two – Introduction to SIP-Session Initiation Protocol 

- SIP Planning – SIP Introduction and Overview

-     SIP Definition – IETF (RFC-3261) and Manufacturers

- CPL-Call Processing Language

- AOR-Address Of Record – q-values

- Location Service – DNS-Domain Name Service

- CPL-Call Processing Language

- B2BUA-Back-2-Back User Agent

-    Session Initiation Protocol for Telephones (SIP-T): RFC 3372

-     SIP-SS7-Signaling System 7 call processing including – IAM-Initial Address Message, Routing label, CIC-Circuit Identification Code and Message Type Code. Examples of Message Type Codes include: Called Number, Calling Number, DPC-Destination Point Code, OPC-Origination Point Code, SS7-ISUP ACM-Address Complete Message, ANM-ANswer Message, CPG-Call ProGress Message, COT-COTinuity Message, SUS-SUSpend Message, RES-RESume Message, FOT-FOrward message Transfer, INR-INformation Request message, INF-INFormation Message, RELease and other messages.

-     SIP – Applications Layer 7 Protocol – Peer-to-Peer protocol

-     SIP – Before and After

-     SIP and Hosted – Better or Worse or Both

-     SIP Signaling – Introduction, URI-Uniform Resource

-     SIP & SBC-Session Border Controllers, servers, gateways,

-     SIP with and without IADs-Integrated Access Devices

-     SIP and SIP Phones, Softphones, Mobility,

-       SIP Signaling Basics – Inbound/Outbound calling

-       UC-SIP Bandwidth Planning – Critical Concepts for PC Video, data and voice

-       SIP Trunking – Four types and counting of SIP Trunking offerings

-       SIP Trunking – Incremental “Slope” Growth

-     CODECS-COmpression-DECompression signal processors – issues and answers

- SIP Trunk Replacement & Disaster Planning

-     SIP & Open Standards

-     SIP and Trunk Replacement – same or different thing

-     SIP and Proxy ARP-Address Resolution Protocol

-     SIP and HSRP-Hot Standby Routing Protocol

-     SIP and MPLS-Multi-Protocol Label Switching – COS and QoS

-     SIP QoS – oxymoron or critical concept

-     SIP on-net and off-net issues – overflow call processing

-     SIP TCO-Total Cost of Ownership – Top-10 Benefits

- SIP Technology – Indepth

-     SIP – OSI-Open Systems Interconnection – “If you do not know where you are going, what difference does it make which path you take”…..Cheshire Cat (Alice in Wonderland)

-                  SIP “Methods” – Writing call processing as easy as email – invite, ACK, bye, etc.

-     SIP Signaling “commands” – 1xx-6xx

-     SIP Inbound and Outbound call processing

-     SDP-Session Description Protocol – headers, Via, Max-Forwards, To:, URL-Uniform Resource Locator, URI-Uniform Resource Identifier, call-ID, Cseq, Contact, Content-Type, Content-Length, Security and others

-     Session Description Protocol Security Descriptions (SDES)

-     SIP Features – Forks, SIP Proxy, Redirect, Presence, Forking – parallel-sequential-mixed, loops, spirals

-     SIP Network devices – UA-User Agent, UAC-User Agent Client, UAS-User Agent Server

-Proxy Server, Redirect Server, Registrar Server, B2BUA-Back-to-Back User Agent

-     SRTP-Secure Real-time Transport Protocol (RFC-3711)

-  Authentication Tag and the Master Key Identifier

-  Encryption

Day Three – Advanced SIP Planning and Security 

- SIP Security – “Best Practices” – Reality Check

-     SIP Security “Best Practices” – overview

-        SIP Firewalls and Security – SPIT-SPam over Internet Telephony, DOS-Denial Of Service, VOMIT-Voice Over Misconfigured Internet Telephony and other emerging problems

-     SIP Security and “Access Policy” – Stateful IP Filtering and Inspection, Static and Stateless IP Filters, TLS-Transport Layer Security, NAT-Network Address Translation, Persistent connection, Multi-homed hosts, etc.

-     SIP and MIM-Man-In-the-Middle attacks – Understanding wireline and WiFi wireless attacks

-     Telephone Numbers – North American Numbering Plan and International ENUM-E.164

- SIP Security Architectures – Building Blocks

-          SIP Security Architectures – eight different VoIP configurations evaluating SIP-Aware Firewalls and other security options:

- Type 1 – Dedicated IP Pipe for VoIP

- Type 2 – Merged MPLS-Pipe with LER Tagging VoIP

- Type 3 – Merged IP pipe with SIP-Aware Firewall (SAFW)

- Type 4 – Separate IP Pipe for VoIP with Existing Non-SIP Firewall and SIP-Aware Firewall (SOFW)

- Type 5 – Merged IP Pipe with Incumbent Non-SIP-aware Firewall, No DMZ Port and SIP-aware Firewall (SAFW)

- Type 6 – Looks like Type 5 but Merged IP Pipe with Incumbent Non-SIP-aware Firewall, No DMZ Port and SIP-aware Firewall

- Type 7 – Merged IP Pipe with Incumbent Non-SIP-aware Firewall with a DMZ Port

-  Type 8 – Merged IP Pipe with Incumbent Non-SIP-aware Firewall

-     Other approaches to SIP Security – Proxy/Gateway Inside the Firewall, Proxy/Gateway in Co-Edge Mode and Proxy/Gateway Outside the Firewall, how Firewalls add time delays to TCP/IP

- 50-Point Comprehensive SIP Security Checklist – more than 50 different security concepts to review and include in the implementation and ongoing network management program

-     SIP Security-Privacy Lifecycle Management – key planning for capturing, storage, users, and disposition (archiving/destruction)

NEW for 2012 – Designing and Planning Scalable SIP Networks

The five areas of discussion presented (with modifications I added) are:

Here are the highlights in text format:

1 – Benefits of Server Virtualization aka Abstraction

- Distributed user endpoints and application servers

- Scalable growth with QoS control abstraction

- MACS* via signaling “abstraction”

*Moves, Adds, Changes & SAS-Stand Alone Survivability – SAS enables backup for SIP devices by the multiple “abstract” local or cloud Media Gateway(s)

2 – Benefits of SIP Trunking

- Session and Signaling control layers for:

- On-net (IP PBX to IP PBX)

- Off-net (IP PBX to local PSTN)

- Off-net (IP PBX to LD/IDDD PSTN)

- SPOC/SMOC-Single/Multiple Point of Connection

- National control points for access (vendors) – Migration path to ALL-SIP & IMS-3GPP

3 – Benefits of Number Transparency

- End-to-end media control (transparency) for voice or other media types

- Access to SS7 applications for text messaging, mobility or call center routing

- Enterprise control over call completion whether PSTN or SIP rather than provider

- Migration path to E.164 & IMS-3GPP

4 – Benefits of SIP Routing

- Eliminates “mesh” network mess

- Add intelligent Session Layer 5 routing – Maintains end-to-end media transparency

- Add media CoS-Classes of Service – Adds SPOC NOC-Network Operations Center

- Migration path to ALL-SIP & IMS-3GPP

5 – Benefits of SIP Proxy Peering Networks

- Platform for multi-vendor services

- Ensures multiple CoS & QoS options

- Migration path to ALL-SIP & IMS-3GPP

- Foundation for any future network needs

- SIP Class of Service & Quality of Service

-     SIP COS-Class Of Service and QoS-Quality of Service – Ethernet meets “smart” IP

-     Managing “real-time” voice with RTCP-Real-Time Control Protocol – MRB-Metrics Report Blocks

-     Inside MRB – what’s what with all the info

- SIP Applications and Future Outlook

-        SIP Applications

– IM-Instant Messaging call screening

- SIP Presence Communicated by IM-Instant Messaging

- Click-to-call and others

-     SIP for Call Centers – calling options and pricing benefits

- Event Notification

- On demand Conferencing

-     Integration of additional “third-party” developed SIP-enhanced services provides additional business and enterprise justification for SIP trunking.

-     UDDI-Universal Description, Discovery and Integration uses standards-based services such as XML, HTTP, SOAP, TCP/IP (define above) uniform service description and service discovery protocol. Discovery services provide a consistent publishing interface and allow programmatic discovery (registration) of services.  Description services provide the means for internet registration – to be found but not advertisement or placement on search engine listings. UDDI file structures are designed using a “publish-once-read-by-many” concept.  That is, web site URL-Uniform Resource Locator can be designed using UDDI standard file structures which can be published to the UDDI server network.  The UDDI network can be accessed (discovered) by search engines, customers and other list builders in a standard published (register) format.  UDDI Registries and protocol servers with:

- White Pages – Names, Address, Contact and Vcard information

- Yellow Pages – Industry categorizations and taxonomies

- Green Pages – Technical information including internal URL file discovery structures

-     UDDI is also designed to replace the robot.txt search engine web site document structure concept. Here are some of the web site description-discovery-registry information retrieved by search engine spiders/bots and other retrieval programs.

-     Voice-driven yellow pages – SALT-Speech Applications Language Tags adds voice commands to web applications.  SALT is an extended set of markup (meta) tags based on XML-eXtensible Markup Language though compatible with HTML-Hyper-Text Markup Language and others.

-     SIP – exciting new applications

-        SIP Total Tutorial with Future Outlook – IMS-IP Multi-media Systems – content servers, wireless integration, media gateways, etc.

- Top-10 Steps to a Successful SIP Implementation

1 – User Needs Assessment

2 – Network Assessment

3 – Systems Upgrade

- Indepth POE-Power Over Ethernet & Comprehensive Disaster Planning Tutorial

4 – Pre–Installation Planning

5 – Data Systems Integration – VLANs, VoWLANS, Planning for WiFi, WiFi and IP Wireless “Roaming,” WiFi Security and more

6 – Installation and Cutover

7 -  Managing Change – Training

8 – Ongoing Use and Expectations

9 – Billing

10 – Managed Services – TCO-total cost of ownership, monitoring, remote support, training, business development and others & Future Applications

- Diagnosing & Tools for Troubleshooting SIP Networks

1 – Problems:

- Delay
- Jitter
- Equipment Configurations
- Clipping & Dipping
- VAD-Voice Activity Detectors -
- Connection Issues
- Echo
- Signal-Noise Level and & Loss
- Comfort Noise
- Packet Loss Concealment

- Zero Insertion

- Waveform substitution

- Model-based methods

- Crosstalk – Nearend and Farend
- Serialization
- Packet Payload Delays
- Packet Sizing Problems – Take the “Vo-eye-P Test” 
- Transcoding Problems
- Asynchronous Transcoding Problems
- Electrical Interference – Surges, Sags, Shared Neutrals

2 – Testing for Problems

- RTCP-XR-MRB-eXtended Reports – Metrics Report Block

3 – More than 30 Problems & Solutions – like “CarTalk” bring your problems to “Nettalk”

4 – Best Practices – review of concepts such as Resiliency & Reliability – QoS in VoIP-SIP

5 – Vendors of Technical Solutions for VoIP Network Management

6 – Conclusions and the Bottom Line

NOTE: Course contents are constantly being update, please inquire about special requirements.

Day Four – (with optional add-on day of labs) – Introduction and Demonstration of Lync Communications Server

Here are just some of the terms used in Microsoft Lync Communications Server.  Each of these terms and others are explained in this Lync Special Edition:

•               Architecture – Front End, AD-Active Directory, Conferencing, Perimeter and Applications Servers
•          PBX Co-existence
•          UC-Unified Communications
•       Advanced Telephony Features
•          PBX Integration Initiative – UC-Compatible, Non-UC-compatible
•          Media Gateway Servers – Forking
•          SIP – Microsoft SIP “Standard” – Inbound/Outbound Calls, OnHold, Making a Call Via a PBX, Answering a Call Via a PBX, Call Forwarding – Always, DND-Do Not Disturb, Call Transfer, 3-Way Conferencing
•          SIP-to-PBX
•          SIP-to-PSTN
•          Microsoft Management Console (MMC)
•          Administrative Tools “Snap-in”
•          Call Detail Records
•          Phone Management and Usage
•          IM-Instant Messaging, Chat, Chatrooms, etc.
•           Video Conferencing Polycom “Roundtable” – Planning and Compliance
•          Saving of IM conversations for compliance
•          Planning for Lync
•          Troubleshooting
•          Trunking – bandwidth, delay, echo
•          Telephone deskset issues
•          IP routing check
•          PBX interoperability
•          Third-party integration – open discussion
•          LCS-Live Communications Server and Lync coexistence

NOTE: This session is constantly changing due to new releases coming from Microsoft such the release of Microsoft Lync.

In addition, there may be “LIVE” demonstration of IM, video conferencing, call processing and other features from a live Lync system.  NOTE: The topics presented may not be the complete list of issues discussed at time permits.  An optional five-day or half-day is available for custom training.

Microsoft is a registered trademark of Microsoft Corporation.

Course Leader – Thomas B. Cross – CEO TECHtionary.com has three decades of experience in startups and consulting advisor with leading providers and venture capital companies in market planning and development, hardware/software design and development, project management, intellectual property in telecommunications, information technology, conferencing, teletraining, telecommuting, groupware, networks, call centers, internet, artificial intelligence and other fields. He has managed the successful development of more than 10 software, hardware and internet products to market and received industry awards for this work. He has authored 13 books, wrote, produced and directed 15 commercial videos and creator and producer of the World’s Largest Animated Knowledge Source on Technology – http://----escape_autolink_uri:081e1145ffa0ed3c87ed0b0373a769b2---- – recipient of Web Hosting Magazine Editors Choice for Best Technical Help.  Tom is a columnist for many leading publications such as Telecom Reseller Newswhere he is the Technology Editor and columnist on Lync Communications Newsletter with a http://crosstalk-techtionary.blogspot.com/.  He is a member of the Technical Board of Advisors for the VoIPSA-VoIP Security Alliance. Tom holds CompTIA Certified Security+, GreenIT and SalesIT Professional certifications and Pearson Vue Certified Test Administrator.   Tom has passed Microsoft Partner Network Sales and Marketing Competency Assessment for Unified Communications.  Tom is also CEO and Managing Editor of Lync User Forum – http://www.lyncuserforum.com.

These are the evaluation scores from Tom’s Presentation to Microsoft Partners & Staff Called “Top-10 Tips for Success” Part 3 of 3 on Telecommunications Networks

• The score is out of 9.0 and that’s the highest score I have ever seen ! Great job Tom – Mike Zeim – Microsoft US Partner Skills Development
• Wow!!!!!!  I didn’t think beating your survey score last time was possible – and look at those results!  Great job, Tom! Jessasym West – Microsoft US Partner Skills Development
•  
• “I personally have sat through a number of Tom’s sessions and found every one of them to be well organized, interactive and informative – OCS Forum and TECHtionary.com are highly recommended resources to bring your organization up to speed on SIP and Microsoft Lync.” – Alan Percy the Director of Market Development at AudioCodes.
•  
• Tom Cross is exceptionally technically astute – the most technically proficient individual I know of in the industry. Yet he has the rare ability to deliver the message in a way the laymen can understand. Tom has done a great job for TBI in the past and he is well respected within the industry. I would recommend Tom for any job or project.”   Geoffrey Shepstone – President – Telecom Brokerage, Inc – 847-353-1842 – Master agency for Qwest, XO, Global Crossing and 30 other companies
•  
• “Tom Cross’ speech on “Why Businesses are Buying VoIP” is certainly one of the top presentations ever given at CTA.  The presentation was insightful, indepth and innovative.  In addition, the presentation was lively, energetic and engaging.  He has the great ability to take complex subjects and make them exciting and understandable. Mr. Cross is a very popular speaker among CTA’s membership and superb communicator.”  Gary L. Witt – Former Executive Director – Colorado Telecommunications Association

Scheduling & Classroom requirements

To confirm delivery dates, fees including estimated travel and expenses are required.  Purchase Orders (P.O.) are accepted but do not confirm dates.  Final expenses are due within 15 business days after completion of class.  Please note that payments received for delivery dates are subject to final approval by Techtionary.  Final student attendance is set 30 days prior to the first day of class with additional student payment due on the first day class.   Techtionary will provide access to the online course at no extra charge.  Client will provide room, computers for lab access, high-quality video projector, desks, power strips, beverages, food service and other amenities.  Classes are conducted each day (subject to mutually agreeable changes) from 0900-1600 with one hour for lunch.  Access to training room will be available from 0800-1700 each day.

Course content and terms are subject to change without notice.  All course content delivery may not be completed due to time restraints such as student questions and special explanations of concepts presented.  Content is wholly-owned by Techtionary Corporation, a Colorado corporation which provides invoice/billing.  Corporate and shipping address is 2525 Arapahoe E-4-313, Boulder, Colorado 80302.