May 2008 Archives

Podcast - VoIP 101 - "How it works"

May 30, 2008 1:15 PM | 0 Comments
Free Tutorials for Channel Partners
Channel Partners can now use multimedia tutorials proven to “reduce the sales cycle” on their website for staff and customer education.  That is, use these free tutorials on the website or email newsletters as a means to help customers understand complex technical subjects. As one channel partner said it best, “these tutorials make it easy for the website to be one of our top sales resources. In addition, having the tutorials on-my-iPod is a cool-tool to show-off to customers.” 
 
This free presentation is produced and presented by TECHtionary.com and is available in Adobe Flash (.swf), audio-only (.mp3), Apple Quicktime (.mov) and iPod/iPhone (.mv4) formats for download at: http://www.bizcastingonline.com/educate/media/podcasts/voip
until June 8 when it will be removed unless you are part of the TECHtionary dealer program where there is no expiration.
 
VoIP 101 - Non-Technical Explanation in 15 Seconds
NOTE: this is an example of one type of system, there will be variations.
 
Get connected to the local router.
Get connected to and phone features from the hosted VoIP provider (or onsite system such as ResponsePoint).
Get connected to IP networks.
Make VoIP/SIP calls via an IP connection.
 
VoIP 101 - "How it works" - Technical explanation in 60 Seconds
 
After the IP phone is plugged into the LAN network and the AC electrical outlet or via POE-Power Over Ethernet cable, the IP phone will connect "bootup" to the services router located on the customer's premise.
 
Now the IP phone downloads from the HSP-Hosted Services Provider (or onsite system such as ResponsePoint) an Image (user features, telephone lines, privileges and other client software to enable service), verifies software version, performs authentication and other network management software as well as Configuration data such as system features, line appearances, corporate and personal telephone directories and other information.
 
Next the IP phone like a PC needs to get an IP-Internet Protocol Address such as (192.30.56.180) from the DHCP-Dynamic Host Configuration Protocol server and connects to the IP network via a DNS-Domain Name Server.
 
Make VoIP/SIP calls via an IP connection.
 
Podcast tutorials, animations and online courses are free to channel partners – see terms and conditions at http://www.techtionary.com/techu/. 
 
For customizing, special discounts, website animations, technical/sales training, technical writing and other services, go to http://www.techtionary.com or please call Tom Cross at 303-594-1694 or cross@gocross.com. 

This is also included in TMC University special course on Microsoft OCS-Office Communications Server at ITexpo.com. For more go here: http://www.tmcnet.com/voip/conference/west-08/tmc-university-microsoft-ocs.htm

If you are a planner, provider, channel partner, or interested party in
Microsoft's ResponsePoint or OCS-Office Communications Server, you are invited to join this new Yahoo! Group. To join, go here:
http://tech.groups.yahoo.com/group/ocs-responsepoint/

Podcast - MPLS-Multi-Protocol Label Switching - Know More, Sell More

May 30, 2008 6:49 AM | 0 Comments
Free Tutorials for Channel Partners
 
Channel Partners can now use multimedia tutorials proven to “reduce the sales cycle” on their website for staff and customer education.  That is, use these free tutorials on the website or email newsletters as a means to help customers understand complex technical subjects. As one channel partner said it best, “these tutorials make it easy for the website to be one of our top sales resources. In addition, having the tutorials on-my-iPod is a cool-tool to show-off to customers.” 
 
This free presentation is produced and presented by TECHtionary.com and is available in Adobe Flash (.swf), audio-only (.mp3), Apple Quicktime (.mov) and iPod/iPhone (.mv4) formats for download at: http://www.bizcastingonline.com/educate/media/podcasts/mpls/
until June 5 when it will be removed unless you are part of the TECHtionary dealer program where there is no expiration.
 
MPLS Tutorial
To begin with, IP-Internet Protocol packets may have a number of labels or "tags" attached to them. MPLS-Multi-Protocol Label Switching is just one type of label. In a Provider Provisioned Virtual Private Network known as PWE3 or PPVPN, there may be more than one label. Here are some terms associated with labeling:
- Push - add a label
- Swap - replace the label
- Pop - remove the label
 
Here are some other terms associated with labeling.
The outer label identifies the LSR-Label Switch Router.
The inner label identifies the destination VPN-Virtual Private Network.
 
Shown in the audio-visual tutorial is the IP-Internet Protocol packet before and with the MPLS “label” attached or “tagged” on as it was originally called. MPLS consists of four elements, label bits, experimental bits, a stack bit and TTL-Time-To-Live bits which indicate the number of Label Switch Routers passed. Shown here is the “multi-protocol” part of MPLS and how it works with the other major networking protocols such as ATM, Frame Relay, Ethernet and others.
 
As if one label was not enough, MPLS providers may add more labels. These labels may exist within the MPLS provider’s network but may be removed or "popped" as they leave the network to the customer premise or "edge" or LER-Label Edge Routers. A PPVPN control module adds "pushes" labels and determines routing via LSR-Label Switch Routers where labels may be "swapped" as they change or cross to other networks called AS-Autonomous Systems.  The term LVC-Label Virtual Channel has been associated with this emerging concept.
 
As long as each MPLS provider or AS-Autonomous System communicates the value of QoS-Quality of Service for the MPLS label to other MPLS providers and routes it accordingly, each carrier can determine their own MPLS labeling system. That is, if each AS carrier routers video as video or email as email or other known rules, then the packets will be treated with the desired QoS. When leaving the MPLS Network or network "edge" the MPLS and other label(s) are popped (off) and the IP packet returns its original size.
 
Background
These audio-video tutorials are available in number of media formats such as Adobe-Flash (.swf), .mp3 (audio only), Apple QuickTime (.mov) - iPod (.mv4) and other formats.  Here are some of the other tutorials available:
-        T-1-PRI-ISDN-Integrated Services Digital Network
-        Hosted VoIP-Voice over Internet Protocol
-        Managed VoIP
-        IAS-CAS-Integrated-Converged Access Services
-        VPN-Virtual Private Networks
-        SIP-Session Initiation Protocol “Event” Notification
-        SIP Security
-        SIP TCO-Total Cost of Ownership
-        SIP Future Outlook – IMS-IP Multimedia Subsystems
-        MPLS-Multi-Protocol Label Switching – “Pushing, Swapping & Popping”
 
Podcast tutorials, animations and online courses are free to channel partners – see terms and conditions at http://www.techtionary.com/techu/. 
 
For customizing, special discounts, website animations, technical/sales training, technical writing and other services, go to http://www.techtionary.com or please call Tom Cross at 303-594-1694 or cross@gocross.com. 
This is also included in TMC University special course on Microsoft OCS-Office Communications Server at ITexpo.com. For more go here:http://www.tmcnet.com/voip/conference/west-08/tmc-university-microsoft-ocs.htm
If you are a planner, provider, channel partner, or interested party in
Microsoft's ResponsePoint or OCS-Office Communications Server, you are
invited to join this new Yahoo! Group. To join, go here:
http://tech.groups.yahoo.com/group/ocs-responsepoint/

Podcast - IAS-Integrated Access Service Morphs into SIP Trunking

May 28, 2008 3:58 PM | 0 Comments
Free Download in iPod/iPhone/Flash/Quicktime/MP3 Formats
 
This free presentation produced and presented by TECHtionary.com is available in Adobe Flash (.swf), audio-only (.mp3), Apple Quicktime (.mov) and iPod/iPhone (.mv4) formats for download at: http://www.bizcastingonline.com/educate/media/podcasts/ until June 4 when it will be removed.
 
Here is the text portion of the presentation.
 
IAD-Integrated Access Devices provide the service known as integrated access also known as converged access (“flex” services) a digital communications service which provides voice telephony connections as well as high speed data communications over the same T-1 circuit. While integrated access service has been around for more than five years, new dynamic bandwidth and other capabilities are now available. While the difference between integrated and converged access is often more marketing than actual, integrated access usually refers to as fixed allocation of bandwidth. For example, the carrier would provide up to 50% of the bandwidth or 12 channels for telephone connections and the remainder or 768 kilobits per second for internet or wide area data networking. 
 
With integrated/converged access, as shown here, bandwidth is not fixed but allocated based on need. Voice is prioritized for high quality of service or QoS. One carrier provides up to 10 high quality and up to 40 low quality voice calls with the remainder of the bandwidth available for data. This means if no one is the phone, the entire T-1 bandwidth is available for data communications. In addition, VoIP-Voice over Internet Protocol, SIP and other features are available depending on specific carrier offerings. 
 
One of the many types of SIP Trunking will be using current IAS-Integrated Access Service also known as CAS-Converged Access Service to support SIP signaling. That is, the provider SBC-Session Border Controller would act as a SIP Proxy server to support SIP UAC-User Agent Client devices, Microsoft OCS-Office Communications Server, softphone clients, analog and other signaling protocols such as FAX and T.38 FAX. QoS is also supported in the LAN via 802.1p/q and on the WAN DSCP-Differentiated Services Code Points or separate MPLS-Multi-Protocol Label Switching. In addition, SIP trunking will be offered via metro/gigabit ethernet. Among the many providers of this technology is Nortel with the CS2100.
 
Background
 
This included in online/onsite courses SIP 2.0c and $499 for OCS-101 Office Communications Server per person (volume and site license discounts available).   Discounts are also available to members of the SIP Forum and MS Partners for $99 per student during May.
 
Courses are free to channel partners – see terms and conditions at http://www.techtionary.com/techu/. 
 
For customizing, special discounts, website animations, technical/sales training, technical writing and other services, go to http://www.techtionary.com or please call Tom Cross at 303-594-1694 or cross@gocross.com. 
 
This is also included in TMC University special course on Microsoft OCS-Office Communications Server at ITexpo.com. For more go here:http://www.tmcnet.com/voip/conference/west-08/tmc-university-microsoft-ocs.htm
 
If you are a planner, provider, channel partner, or interested party in
Microsoft's ResponsePoint or OCS-Office Communications Server, you are
invited to join this new Yahoo! Group. To join, go here:
http://tech.groups.yahoo.com/group/ocs-responsepoint/

Fundamentals of Bandwidth - An Introductory Tutorial

May 28, 2008 4:25 AM | 0 Comments
This is a tutorial on the fundamentals of bandwidth from transmission of sound waves and analog to digital, advanced optical fiber optics and the concept of data "packet" transmission. Voice is a series of waves which your ear analyzes in terms high and low tones such as a high-pitched soprano voice or low-pitched baritone. These waves of highs and lows are called a sine wave. The top of the sine wave is a 1 and the bottom a 0. Analog is a varying sine 1/0 wave signals of different frequencies (FM) and/or amplitude (AM). Analog is used in residential telephone service called POTS-Plain Old Telephone Service and small business telephone service called 1FB or business lines and trunks. While analog transmission may have many values, digital transmission consists only of 1s and 0s or bits. If you prefer, digital is like a light switch - on or off. Bandwidth is refers to the number of bits transmitted over one time interval or second. 
Bandwidth determines how much information (voice, data, video) of any kind, can be sent to another location at any given time and how fast that information can get there.   There are many types of digital transmission. One kind of digital transmission is called TDM-Time Division Multiplexing where transmission is divided into time-divided fixed length channels. The PSTN-Public Switched Telephone Network is a network of TDM-Time Division Multiplexed circuits such as: T-1 or DS-1 is 1,544,000 bits per second of 24 channels of 56,000 for voice and data and 8,000 for signaling. ISDN-PRI-Primary Rate Interface is also 1,544,000 bits per second but organized 23 B-Channels of 64,000 BPS for voice/data and 1 D-Channel of 64,000 BPS for signaling. T-1-ISDN-PRI is "provisioned" (installed) using two-pair twisted-copper wiring (referred to as a 4-wire circuit). However, optical fiber is used with optical pulses and multiple-colored "dense waves" to 10-40 gigabits per second and increasing speeds. While bandwidth is a "stream" of data, groups of 1s and 0s are organized into groups of eight called a byte or octet. Bytes are grouped into packages called packets to perform tasks. There are hundreds of types of packets. Many packets become accepted or standardized by different industries and then referred to as a protocol. For example, IP-Internet Protocol is used to send email, data or voice.
More Details:
In order to take analog audio to digital a CODEC is required. The concept of a CODEC is an important part of VoIP/SIP. While explained in detail later, the key point of a CODEC is to take noise, music and other forms of audio and transform it into a digital format. Coding-encoding-decoding is the process of sampling quantities and putting them into digital values of voice, music or other sounds. The Nyquist-Shannon sampling theorem states that the sampling frequency must be at least twice as high as the highest input frequency for the result to closely resemble the original signal. A 4,000 Hz-Hertz voice pattern would be sampled at 8,000 BPS-Bits Per Second. Organized into 24 separate voice channels with spacing bits (called Framing bits) separating each 24 segments of 8,000 bits becomes a T-1-Transmission Level One transmission circuit of 1,544,000 BPS. For example, in MP3-Motion Picture Experts Groups Version 3 different compression (sampling or quantizing) rates are needed for different music quality levels such as 128 KBPS - CD quality (twice normal bandwidth), 96 KBPS - near-CD quality and 64 KBPS - FM radio quality.

This is included in online/onsite courses SIP 2.0c and $499 for OCS-101 Office Communications Server per person (volume and site license discounts available).  
 
*Courses are free to channel partners – see terms and conditions at http://www.techtionary.com/techu/. Discounts are also available to members of the SIP Forum and MS Partners for $99 per student during May. For customizing, special discounts, website animations, technical/sales training, technical writing and other services, go to http://www.techtionary.com or please call Tom Cross at 303-594-1694 or cross@gocross.com. 

This is also included in TMC University special course on Microsoft OCS-Office Communications Server at ITexpo.com. For more go here:http://www.tmcnet.com/voip/conference/west-08/tmc-university-microsoft-ocs.htm

New Yahoo Group on Microsoft ResponsePoint & OCS-Office Communications Server

May 27, 2008 7:49 AM | 0 Comments
If you are a planner, provider, channel partner, or interested party in
Microsoft's ResponsePoint or OCS-Office Communications Server, you are
invited to join this new Yahoo! Group. To join, go here:

http://tech.groups.yahoo.com/group/ocs-responsepoint/

“You Can’t Sell What You Don’t Know” – Selling SIP Systems & Services

May 22, 2008 9:52 AM | 0 Comments
Top 10 Critical Issues in Selling and Implementing SIP Systems & SIP Trunking 

Included in Free* SIP Course for MS and all other Channel Partners

By Thomas B. Cross CSSP – CEO TECHtionary.com

The panelists on the session Selling SIP Trunking at recent Channel Conference missed the mark in helping the SRO audience get to square one for selling SIP Trunking.  Without going into all the issues they could have and should have addressed, here’s are three tips.  

1 - What is SIP?  Make sure you know what SIP means.  It means Session Initiation Protocol, not anything else.  Basically, SIP provides signaling, like car traffic lights, in order that SIP devices can call other SIP devices over a symmetrical broadband internet connection (no ADSL).  If you want to know more about the working of SIP protocol, get involved in technical discussions or your product interoperability compliant, go to the SIP Forum, a nonprofit industry interoperability organization at www.sipforum.org. The SIP Forum will help you understand the industry, players, protocols such as RTP-Realtime Transport Protocol, SDP-Session Description Protocol and others as well as RFC-Request For Comments that are the basis for all SIP development.


2 – SIP devices can be hardphones, wireless phones, softphones (software) and other devices such as soda machines and in the future nearly every other device.  SIP moves the “intelligence” from the PBX/CO into the device.  That is, SIP devices communicate directly with one another without the need for a PBX or CO-Central Office switching system.  This is just like the way your PC communicates directly with a website.  This means the features are in the SIP device, not PBX.  Practically speaking this means I can use my laptop with softphone software as a telephone and can take it anywhere and plug in to an internet connection and begin making outgoing or receiving incoming calls from other SIP devices without a PBX.  If I need to call outside my SIP network or receive a call, my SIP gateway provider (in this case www.simplesignal.com) gives me a PSTN number which you can call and no matter where I am you can call me.  Features such as voice mail, transfer, conference, etc. can be added through software from the SIP system or SIP gateway provider.

3 – Bandwidth planning is paramount.  SIP devices use a CODEC (coder-decoder, compression-decompression), a technical term for computer chip, to process calls into international standard voice formats.  One major CODEC is G.711 provides for high-performance “toll-quality” calls and uses 64 KBPS per call.  A low-performance CODEC (much like cellular service) for low-bandwidth voice calls of 8 KBPS is G.729.  There are other CODECS supported by various manufacturers. Check specific companies for details. 
The most important point is that in planning for SIP implementations allocate 80-100 KBPS per call for G.711 and around 30 KBPS per call for G.729.  That is, while G.711 uses 64KBPS of voice it needs more bandwidth because of the packetizing (RTP-TCP/UDP-IP overhead) for an internet protocol network.  Here’s an easy rule of thumb, for G.711 take the total number of simultaneous (concurrent) calls times 100 KBPS and that is the bandwidth the customer needs for peak “busy hour” times.  In addition, SIP trunking providers will limit the number of voice calls based on the CODECS they support.   One SIP trunking provider supports 11 calls using G.711 and 42 calls with G.729. However, the customer benefit when users are not on the telephone with the bandwidth is automatically or “dynamically” available for their data needs. In other words, check with your SIP trunking provider, media gateway manufacturer and other “parts” in the network. That is, YMMV-your mileage may certainly vary.  

There are seven other critical concepts such as security, interoperability, pre-installation planning, data systems integration and others you need to be “SIP smart” in selling SIP that are included in OCS-101 and SIP Essentials 2.0c available in the onsite and online courses. 
 
The online version is $299 for SIP 2.0c and $499 for OCS-101 Office Communications Server per person (volume and site license discounts available).   Discounts are also available to members of the SIP Forum and MS Partners for $99 per student during May.
 
For customizing, special discounts, website animations, technical/sales training, technical writing and other services, go to http://www.techtionary.com or please call Tom Cross at 303-594-1694 or cross@gocross.com. 
 
*Courses are free to channel partners – see terms and conditions at http://www.techtionary.com/techu/. 
 

Microsoft Talks SIP

May 21, 2008 12:27 PM | 0 Comments
A great read on Microsoft's view on SIP:

'SIP Trunking' in Office Communications Server

In the Office Communications Group (OCG) white paper “Integrating Telephony with Office Communications Server 2007” we stated that “SIP Trunking” was out of scope for that release but was “under consideration” for future releases.  We are always reluctant to commit features to releases in advance of the official release announcement; however it is reasonable to say that the process of consideration is underway.

http://communicationsserverteam.com/archive/2008/05/20/168.aspx

Improve Your Marketing Image

May 21, 2008 6:54 AM | 0 Comments
Need help in training, technical writing, creating ‘award-winning’ customer case studies, webseminars, podcasts, customer presentations, effective web tutorials, SEM, PR and other services, contact Tom Cross at 303-594-1694, cross@gocross.com or http://www.techtionary.com.

SIP Tips – How SIP Works Tutorial

May 20, 2008 12:38 PM | 0 Comments
Since too few people really know how SIP works, here is the text explanation of each step of the way. After the IP phone is plugged into the LAN network and powers up via the electrical outlet or via POE-Power Over Ethernet cable, the phone needs to get an IP-Internet Protocol Address from the DHCP-Dynamic Host Configuration Protocol server.

SIP Signaling
User dials number 303-594-1694. SIP-URI-Uniform Resource Identifier is retrieved from DNS-Domain Name System: sip:tom@xyz.com;Transport=UDP. 
The SIP INVITE along with the SDP-Session Description Protocol is formatted into an Internet Message Format and encapsulated into ethernet and sent via the LAN Switch to the router and encapsulated into IP or IP-MPLS and UDP and the SIP INVITE is sent via TCP, UDP and other protocols to the destination proxy. The caller receives a 100 (Trying) response indicates that the INVITE has been received and that the proxy is working "on behalf of" to route the INVITE to the destination. The caller receives a 180 (Ringing) and begins ear ringing using an audio ringback tone or by displaying a message on the telephone screen.
When the person called picks up the handset, the SIP phone sends a 200 (OK) response to indicate that the call has been answered. The 200 (OK) message contains a SDP-Session Description Protocol media description of the type of session that the other party is willing to establish. An ACK message is sent.
SIP Media Session
User begins talking. Voice is transcoded into a CODEC, e.g. G.711, G.729 based on SDP. Voice is packetized. RTP is added and encapsulated in Ethernet and sent via the LAN Switch to the router and encapsulated into IP or IP-MPLS and UDP. If on-net IP network, the digital data is converted to an optical data stream and sent via optical fiber to an internet or internal router. If off-net, the digital data is channelized using a MG-Media Gateway to a TDM-Time Division Multiplexed channel and sent via an optical data stream to a Class 5 CO-Central Office switch for connections to the PSTN-Public Switched Telephone Network. User hangs up and a BYE message is sent. A 200 ACK is received confirming disconnect.
 
If you want to know more, this information is part of OCS-101 and SIP Essentials 2.0c courses available onsite and online.  The online version is $299 for SIP 2.0c and $499 for OCS-101 Office Communications Server per person (volume and site license discounts available).   For more information go to http://www.techtionary.com or please call Tom Cross at 303-594-1694 or cross@gocross.com.  Discounts are also available to members of the SIP Forum and MS Partners for $99 per student during May.
Courses are free to channel partners – see terms and conditions at http://www.techtionary.com/techu/

Software+Services – Hosted (Web-based) Exchange Mail & OCS Services

May 20, 2008 9:36 AM | 0 Comments
Referred by Microsoft as “software+services” or as others call it SAAS-Software As A Service and more commonly referred to as hosted or web-based applications, there are more than 100 companies who offer hosted Microsoft Exchange email services. In addition, a growing number of these ISP’s offer Microsoft OCS-Office Communications Server telephony as an additional featureset. Microsoft Hosted Exchange Filtering provides both inbound and outbound anti-spam, anti-virus, policy, disaster recovery along with administrative console, junk mail quarantine and other features. In addition, Hosted Exchange integrates Microsoft Forefront security software with SharePoint.  SharePoint is what was called computer-conferencing, groupware and other names. It’s a great concept developed in the 1970s by a number of different organizations including such notables as Douglas Engelbart – the Father of the Mouse - http://en.wikipedia.org/wiki/Douglas_Engelbart.
 
 
Email archiving is critical for legal discovery and proper corporate governance. Hosted Exchange Archive provides text indexing of e-mail and attachments, IM-Instant Messages and other documents. In addition, government compliance tools for supervising, escalating, and tracking messages are available. With Hosted Exchange, users can access quarantined mail and Administrators can perform message administration. Either party can access mail archive via the Web for real time mail functionality. In addition, archive repository benefits from upstream spam and virus protection features. Here as you can see Hosted Exchange reduces and may eliminate a substantial number of onsite equipment requirements while greatly enhance an organization's ability to support multi-site or new locations. Traditional Exchange Server 2007 "on-premise" implementations remain viable for single location organizations where DMZ-De-Militarized Zone functions are required for HTTP, ecommerce and other applications including OCS-Office Communications Server. In addition, situations where internal locations are behind the DMZ would be appropriate for an on-premise approach.
 
If you want to know more, this information is part of OCS-101 and SIP Essentials 2.0c courses available onsite and online.  The online version is $299 for SIP 2.0c and $499 for OCS-101 Office Communications Server per person (volume and site license discounts available).   For more information go to http://www.techtionary.com or please call Tom Cross at 303-594-1694 or cross@gocross.com.  Discounts are also available to members of the SIP Forum and MS Partners for $99 per student during May.
Courses are free to channel partners – see terms and conditions at http://www.techtionary.com/techu/

Bi-Polar Disorder - Privacy-Security Lifecycle Management

May 20, 2008 9:24 AM | 0 Comments
Bi-Polar Disorder - Privacy-Security Lifecycle Management
Balancing Issues in World of Telepresence
New Roles for Authors – Actualizers - Auditors – Analyzers
There is an increasing amount of information available on the “bipolar” disorder of security and privacy. It seems bipolar for a number of reasons: most people are fiercely protective of their privacy but don’t want to have to deal with security roadblocks; management wants and needs to protect customer privacy but not at an exorbitant cost; and the line must be drawn between where security for company information ends and security for customer information begins. So this is one of those issues where there are no "right answers," just practical uses based on organizational needs and management commitment. Many of the "people issues" are driven by policy and by the needs of management, with the results (good or bad) blamed on them.
 
With that in mind, here is a policy presentation rather than a definitive security-privacy plan. We have found that, from an extensive review of the current writings on SOX, HIPAA and other regulatory/judicial findings as well as interviews with leading security experts, there should be four key players "holding the stool up,” in any good security arrangement: authors, actualizers, auditors, and analyzers.
 
Before anyone does anything the issues need to be laid out. In this animated tutorial, let’s start with the technical ingredients of any secure data depository. First, data needs to be captured or collected. Here are some basic types of data – in person or via people, media types – audio, paper, video, machines – ATM, POS-point of sale, POP-point of purpose, game, manufacturing, remote diagnostics, outside sources such as databases, research and other studies and online – via search engines, email, IM and other retrieval or interactive sources. Second, what type of container or storage media would the data be placed in. Examples of storage would be on paper, online, SANs-storage area networks and real-time archive. Third, how and by whom is the information going to be used? Examples would be users, customers – includes channel partners and mashed up. Mashed information is a new category not mentioned by most sources today as it’s a relatively new phenomenon. The concept of SAAS-software as a service more commonly known as web/hosted applications is more than a single application used by itself. More and more applications offer the customer integration with other applications. CRM-customer relationship management is merged with VoIP/SIP, video is merged with geographic mapping and presence is included with meetings to name a few simple ones. Mashed applications are more problematic in that other providers may likely not agree to your security/privacy requirements or you to their capabilities. Compliance and certification such as ISO and ITIL take on a whole new dimension. Fourth, disposition has three major types; archive for compliance, uncertain and pending/destroyed. We have reviewed other approaches and you should certainly have your own lifecycle specific to your enterprise but this is included as a start. Next comes the people part of the equation. Concepts relating to the technology can be easily explained, but sometimes it is the human interface that may not be so easily understood or resolved.
 
First, Authors are the senior/executive management leaders providing strategic direction in the form of all-encompassing ideology. Second, Actualizers include anyone and everyone who touches data, applications, systems, managers, archivists and anyone else. Third, Auditors may have the simplest role, that of checking on how well actualizers follow authors’ policies. Fourth, the Analyzer reviews and checks the auditors to determine if the auditors are practicing their processes and documentation uniformly and universally across the enterprise. In addition, given increasing levels of compliance and regulatory oversight, the Analyzers provide an additional independent layer of review and analysis.   This additional layer of review is becoming more and more necessary because the processing of balancing security and privacy is getting more, not less complex.
 
As with any bipolar disorder security experts and privacy advocates need to find a common ground for discussion especially as voice (VoIP/SIP), video and presence (a/k/a telepresence) is becoming mainstream. However, like in the last scene of the great sci-fi movie “The Day the Earth Stood Still, “the test of any such higher authority (security) is, of course, the police force that supports it.” It’s great to have policies but if there is no police force to enforce them and evaluate that enforcement there is no true security or privacy for all, at all.
 
If you want to know more, this information is part of OCS-101 and SIP Essentials 2.0c courses available onsite and online.  The online version is $299 for SIP 2.0c and $499 for OCS-101 Office Communications Server per person (volume and site license discounts available).   For more information go to http://www.techtionary.com or please call Tom Cross at 303-594-1694 or cross@gocross.com.  Discounts are also available to members of the SIP Forum and MS Partners for $99 per student during May.
Courses are free to channel partners – see terms and conditions at http://www.techtionary.com/techu/
 

Cisco Unified Communications Manager Denial of Service Vulnerabilities

May 20, 2008 9:19 AM | 0 Comments
May 16, 2008
If you look below this notice, you will see what I published yesterday on Security-Privacy, so what comes from Cisco should be no surprise.  Cisco Unified Communications Manager, formerly Cisco CallManager, contains multiple denial of service (DoS) vulnerabilities that may cause an interruption in voice services, if exploited. These vulnerabilities were discovered internally by Cisco. The following Cisco Unified Communications Manager services are affected:
  • Certificate Trust List (CTL) Provider
  • Certificate Authority Proxy Function (CAPF)
  • Session Initiation Protocol (SIP)
  • Simple Network Management Protocol (SNMP) Trap
For those of you unfamiliar with certificates, here’s a quick tutorial. 

Cisco also has unique names for some of their certificate functions. For more on this alert go here:
http://www.cisco.com/warp/public/707/cisco-sa-20080514-cucmdos.shtml

Free Training for Channel Partners

May 12, 2008 1:41 PM | 0 Comments
TECHtionary.com Announces Free Training for Channel Partners
on SIP, OCS-Office Communications Server, Digital Communications Courses
Courses Proven to Reduce Sales Costs & Buy Cycle
TECHtionary.com today announced new program to help channel partners (independent, agents, VARs, dealers, brokers, interconnect) grasp new technologies faster and increase revenues faster. These free courses “gives access to channel partners who are time-constrained, mobility-driven and performance-pressured in today’s ruthless business world. Channel professionals can also learn on the go and anytime they want to whether waiting at the airport or a customer’s office, driving down the street or riding on the bus/train/plane,” Cross added. 

Courses are free for agents of TECHtionary-TBI-Telecom Brokerage Inc. where agents not only receive free training but commissions on network services delivered to customers.  For more information on this program, click here: http://www.tbicom.com/tbiu/.  There is no limit to the number of employees who can attend the classes though each employee must be registered with TBI.   TBI-Telecom Brokerage Inc, is one of the largest master agency in the U.S. TBI represents all of the leading network services providers with a complete set of Local, LD, Internet, Data and advanced telephony SIP products. According to Geoff Shepstone, CEO of TBI, “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. These courses can bring immediate returns to channel partners.”

Course titles include:
- CTM-Communications Technology Manager -
Introduction to the Fundamentals of Digital Communications Technology
- WTM-Wireless Technology Manager
Job Training and Implementation of WiFi, WiMax, Cellular and IMS
- VBE - VoIP Business Executive 
Channel Provider & Partner Business Sales & Technical Strategies for VoIP-SIP
- STE - SIP Essentials - Technology Business Executive 
SIP Essentials - Comprehensive guide SIP-Session Initiation Protocol
- NDP-Network Design Professional
Planning for optical, routing, routing protocols and preparation for industry-wide Cisco, Microsoft, Avaya, Comptia and other certifications  
- OCS-Office Communications Server Complete
Indepth explanation of Microsoft OCS-Office Communications Server  
- Media Library
Audio-video tutorials are available in number of media formats such as Adobe-Flash (.swf), .mp3 (audio only), iPod (.m4v) and Apple QuickTime (.mov). 
 
For more information please call Tom Cross at 303-594-1694 or cross@gocross.com.

Voice Peering Fabric – The Future of SIP Interconnections

May 6, 2008 3:46 PM | 0 Comments
Imagine that your corporate ethernet LAN-Local Area Network and any other Fortune 500 company LAN can connect via a Voice Peering Fabric (www.thevpf.com), a service of Stealth Communications. Calls from any LAN business or otherwise can be routed through the VPF and then to your LAN and to your desk. Look Ma, no Internet routing hops would be involved. Any SIP provider or Enterprise can setup peering through this system via ethernet. The calls would cost nothing per minute, aside from the fixed monthly cost of the digital link.   This concept just extends LAN-to-LAN functionality to almost anywhere. VPF is accessible at major carrier hotels in North America and London, UK.  If an Enterprise or Service Provider is not located at a common location as the VPF, they can provision Ethernet connectivity from their location to the nearest VPF location.

According to Shrihari Pandit, CEO of the Stealth Communications, “we created a program called the VPF Carrier Alliance to assist organizations in obtaining competitive Ethernet connectivity into the VPF. The alliance is composed of regional, national and international carriers that specialize in providing Ethernet connectivity. Once an organization is connected into the VPF, they are able to establish SIP peering with other connected organizations.”   Members of the VPF use the service to buy and sell voice origination and termination services, as well as ASP services while providing complete transparency. The service allows organizations to instantly identify all available networks and services at the most competitive rates. Members then get connected to these networks and services within minutes by using the "fabric" component of the VPF. In addition, organizations can access the VPF ENUM and SRV Registry, as a toll bypass system, to send and receive free telephone calls across the VPF with other users of the registries. Here are some of the services offered by members of the VPF:
- SIP/H.323 Origination and Termination Trunks
- Operator Service, Directory Assistance & E911 Routing and
- Telco Database Services: 8XX, CNAM, LIDB, LNP.

In the case a company wants to exchange voice traffic with another company that is not on the VPF, they can either peer with that company over another network (such as the public Internet) or route that call through the PSTN (Public Switched Telephone Network.)  In the event the call needs to be routed to the PSTN, there are carriers on the VPF that can route such traffic to the PSTN on their behalf (bilateral relationship.) VPF supports bilateral and multilateral peering relationships.   Bilateral peering is a term that describes when two parties decide to exchange traffic on a settlement basis. For example, there may be a cost per minute to send calls. Multilateral peering enables a community of companies to exchange traffic for free (without settlement) between members of the community. Multilateral peering typically uses ENUM (Electronic Number Mapping) that maps telephone numbers to Internet addresses and is based Internet DNS technology.  For more go here: http://www.thevpf.com/
 
If you want some real “live” peering, plan to attend the Voice Peering Forum, a biannual conference, scheduled for June 23-24 2008 at Hotel Nikko in San Francisco which brings together over one hundred unique organizations from all segments of the information technology and telecommunications industry to network and discuss the latest in peering, routing and interconnection of networks and the applications they support. Go here to get more http://www.voicepeeringforum.com/

If you want to know more, this information is part of OCS-101 and SIP Essentials 2.0c courses available onsite and online. The online version is $299 for SIP 2.0c and $499 for OCS-101 Office Communications Server per person (volume and site license discounts available).   For more information go to http://www.techtionary.com or please call Tom Cross at 303-594-1694 or cross@gocross.com. 
Discounts are also available to members of the SIP Forum and MS Partners for $99 per student during May.
 

OCS Exposed - SQL-Structured Query Language for Telephony

May 6, 2008 12:30 PM | 0 Comments


OCS Exposed - SQL-Structured Query Language for Telephony

An internal telephony user can connect to another internal telephony user by using the Office Communicator 2007 client. A user can initiate a call to another user by either selecting the user from a contacts lookup list or dialing that user’s contact number. When the user initiates a call using a SIP-Session Initiation Protocol client, the client sends an SIP "Invite" message along with the SIP URI-Uniform Resource Indicator of the call recipient to the Front End Server. After receiving the SIP Invite, the Front End Server queries the SQL Server 2005 database to check if the SIP URI for the call recipient is present in the database. If the SIP URI is available, the Front End Server applies client rules on the Invite, and then routes the Invite to all active SIP clients corresponding to the URI of the call recipient. If the Front End Server receives a busy or does not receive an acknowledgement from any one of the SIP clients of the call recipient, the Front End Server then routes the call to the Exchange Server 2007 voice mail service. 

The voice mail service generates an e-mail to the call recipient along with the voice mail attachment as an audio file. However, if an acknowledgement is received from any one of the SIP clients of the call recipient, the Front End Server responds with a Ringing message to that client. The Front End Server also sends a Cancel Invite message to all the other registered clients of the call recipient. A call session is established when the call recipient answers the call by using the SIP client. The Front End Server then opens a media stream between the clients of both users. After the conversation, either of the clients can send a Bye message and the Front End Server terminates the call session.
 
The Front End Server also indexes the SQL Server 2005 database to translate (map-resolve) the normalized (canonical) number to a user URI-Uniform Resource Indicator. Note: URI refers to the complete SIP telephone address not just the mail URL-Uniform Resource Locator. If the Front End Server does not receive an acknowledgement from any one of the SIP clients of the call recipient, the configured InfoAgent or Outbound Router logic running on the Front End Server detects that the call recipient is not answering the call. 
 
For those unfamiliar with SQL, SQL-Structured Query Language which is an ANSI-American National Standards Institute standard computer language for accessing and manipulating database systems. SQL statements are used to retrieve and update data in a database.   SQL Statements contain "same type" information (e.g. store address, SKU, shelf count) sometimes referred to as columns. The DML-Data Manipulation Language is used to retrieve, add/insert and change/modify database information.  
For a detailed animated tutorial on SQL, database concepts and other topics, go to http://www.techtionary.com
 
If you want to know more, this information is part of OCS-101 and SIP Essentials 2.0c courses available onsite and online. The online version is $299 for SIP 2.0c and $499 for OCS-101 Office Communications Server per person (volume and site license discounts available).   For more information go to http://www.techtionary.com or please call Tom Cross at 303-594-1694 or cross@gocross.com. 
Discounts are also available to members of the SIP Forum and Microsoft-MS Partners.




MPLS-Multi-Protocol Label Switching Multimedia Presentation

May 2, 2008 7:44 AM | 0 Comments
To begin with, IP-Internet Protocol packets may have a number of labels or "tags" attached to them. MPLS-Multi-Protocol Label Switching is just one type of label. In a Provider Provisioned Virtual Private Network known as PWE3 or PPVPN, there may be more than one label. Here are some terms associated with labeling:
- Push - add a label
- Swap - replace the label
- Pop - remove the label
 
Here are some other terms associated with labeling.
The outer label identifies the LSR-Label Switch Router.
The inner label identifies the destination VPN-Virtual Private Network.
 
Shown in the audio-visual tutorial is the IP-Internet Protocol packet before and with the MPLS “label” attached or “tagged” on as it was originally called. MPLS consists of four elements, label bits, experimental bits, a stack bit and TTL-Time-To-Live bits which indicate the number of Label Switch Routers passed. Shown here is the “multi-protocol” part of MPLS and how it works with the other major networking protocols such as ATM, Frame Relay, Ethernet and others.
 
As if one label was not enough, MPLS providers may add more labels. These labels may exist within the MPLS provider’s network but may be removed or "popped" as they leave the network to the customer premise or "edge" or LER-Label Edge Routers. A PPVPN control module adds "pushes" labels and determines routing via LSR-Label Switch Routers where labels may be "swapped" as they change or cross to other networks called AS-Autonomous Systems. The term LVC-Label Virtual Channel has been associated with this emerging concept.
 
As long as each MPLS provider or AS-Autonomous System communicates the value of QoS-Quality of Service for the MPLS label to other MPLS providers and routes it accordingly, each carrier can determine their own MPLS labeling system. That is, if each AS carrier routers video as video or email as email or other known rules, then the packets will be treated with the desired QoS. When leaving the MPLS Network or network "edge" the MPLS and other label(s) 
are popped (off) and the IP packet returns its original size.
 
These audio-video tutorials are available in number of media formats such as Adobe-Flash (.swf), .mp3 (audio only), Apple QuickTime (.mov) - iPod (.mv4) and other formats.  For more information and scheduling, please call Tom Cross at 303-594-1694 or cross@gocross.com.
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