Effective
1-1-2015
Independent Course Program Evaluations
"Certification is
demonstrating to existing and prospective customer's your commitment
to excellence. The Certified SIP
Professional Program sets a new standard for business technology education
excellence.” Doug Green Publisher UserNews
"CSP provides the certification--the key to
success today to increase sales and improve support. Certification is
essential." Bob Speers - CEO Speers Telecom
“CSP certification gets everyone on the same page faster, resulting in more qualified sales, faster deployments and better staff/channel collaboration." Matt Jolly – Senior Engineer Vology
100+ Compelling Reasons for Achieving SIP Certification -
click on image for article
University
For access to 3,000+ animated
tutorials, click here: http://techtionary.com
Summer
Certification Dates
Discounts
for persons from .edu and nonprofits are available. Email for special
invoicing.
Topic 457 - Tuition and
Fees Deduction - You may be able to deduct qualified tuition and related
expenses that you pay for yourself, your spouse, or a dependent, as a tuition
and fees deduction. Or, 1)
an American Opportunity credit or Lifetime Learning credit; or (2) if
applicable, a business expense, which is generally an itemized deduction. Click on image
for IRS ruling.
Certified SIP Professional (CSP) Program 2015
1-3
Days Onsite or Webseminar
NEW for 2015
- Designing and Planning Scalable SIP Networks
©
TECHtionary Corporation – all rights reserved - Made in USA
For more information and scheduling, please call Tom Cross
303-594-1694
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, “CSP 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 onsite 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 certification 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 SIP 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 certification 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
onsite/webinar vendor-neutral 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 high performance
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 Hosted, Cloud, 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, Fuzzing, 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 digital
telecommunications.
Quick Overview
NOTE: All course content delivery may not be
completed during the course delivery due to time restraints such as student
questions and special explanations of concepts presented.
Fundamentals of Data/Internet Telecommunications - This is for
inhouse training.
- 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
Certified
SIP Professional (CSP) Program
Day One - 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
Certified
SIP Professional (CSP) Program
Day Two - Advanced SIP Planning and Security
20 Types of SIP Security Attacks -
Malformed message attacks, SPIT-SPam over Internet Telephony,
Robocalling SPIT, Memory "leak" buffer overflow attacks, VOMIT-Voice
Over Misconfigured Telephony, DOS-Denial-of Service attacks (overload, part
SYN, FIN), TDoS - Telephony Denial of Service attack, Eavesdropping,
Masquerading (Trojan Horses), Media injection, SIP redirect, Endpoint viruses,
malware, spyware, Calljacking - hijacking, RTP/RTCP session teardown attacks,
RTP/RTCP malformed messages, RTP/RTCP buffer overflow attacks, RTP play-out or
media spamming, SDP changing CODECs attacks, malicious RTP packets, and others
-
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)
A few tools
for SIP Security - Sniffing Tools, Scanning and Enumeration Tools, Packet
Creation and Flooding Tools, Fuzzing Tools, Signaling Manipulation Tools, Media
Manipulation Tools
NEW for 2015
- 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
- New -
SIP-SIP InterCall Working - SIP-I to SIP and H.323
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 and "Push" Notifications
- 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 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 Future
Outlook
– IMS-IP Multi-media Systems –
content servers, wireless integration, media gateways, etc.
- New Detailed IMS
architecture scenarios
- WebRTC and HTML5 Internet-centric
communications environment
Here are six WebRTC “solutions scenarios.” Of course your
situation may be different, however, what I was most excited about what how
WebRTC can seamlessly work with SIP in IP-PBX environments such as Avaya,
Cisco, ShoreTel and Lync. This gives WebRTC an “overlay” approach rather rip
and replace. This is just the first part of many more discussion on this
rapidly evolving technology.
#1 Enterprise
Architecture
#2 Enterprise
Architecture
#3 – Multiple
Backend Systems
#4 – Cloud
Provider
#5 – Lync
Integration
#6 – PSTN Off
Net
- Mobile SIP apps - review of
unified communications apps
- 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
- SIP business architectures -
Premise, Managed, Hosted, Cloud
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
- Qos Management
- Peak Call Bursting
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.
Course
Leader - Thomas B.
Cross, B.S. M.S-Telecommunications – 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://www.techtionary.com – 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 Cloud, Unified
Communications and Lync User Forum 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 Cloud-UC Forum – http://ucomapps.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
|
|
Instructor |
Courseware |
Overall Satisfaction |
Business Results |
Learning Effectiveness |
Job Impact |
Environment |
Average |
|
Average |
8.73 |
8.38 |
8.73 |
8.64 |
8.45 |
8.45 |
8.91 |
8.59 |
•
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
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 (credit cards accepted). 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,
if applicable on the first day class.
Techtionary will provide access to the online course at no extra charge
within five (5) days at the end of the onsite course. That is, the online version is only available
to students who attend the onsite course. Client will provide room, computers
for lab access, high-quality video projector and screen, 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 during the course delivery 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.