August 2008 Archives

New Comprehensive Planning, Implementation, Security, Troubleshooting Guide for
UC-Unified Communications, SIP, OCS-Office Communications Server Networks

Course Also Available Onsite-Anywhere, Online 7x24 and Via Webseminar

PORTLAND - August 28 - Walt Medak Associates today announced its "SIP Planning Guide Course," onsite courses for SIP-VoIP, UC-Unified Communications, Microsoft OCS-Office Communications Server & Response Point and IPT-Internet Protocol Telephony networks.

"The SIP Planning Guide course expands our Avaya Definity classes to the critically important next level," noted Walt Medak CEO. "SIP is a complex process because SIP is not just a single location service but one that encompasses nearly all aspects of business communications and computing," Medak added.

The SIP Planning Guide course provides the means for customers who are planning, implementing and expanding their communications networks to "get SIP smart" with guidance, ideas and tools. In addition, this course is designed to provide manufacturers, providers, VARs, dealers, agents, analysts and others with new insights into SIP. With more than one hundred new concepts for review, the SIP Planning Guide course is vital to anyone doing UC/VoIP/SIP/IPT for network solutions and premise-based implementations.

Some of the key highlights in SIP Planning Guide course:
- VLPS-Virtual Private LAN Service with MPLS
- Privacy-Security Lifecycle Management - Authors - Actualizers - Auditors - Analyzers
- PLC-Packet Loss Concealment - zero Insertion, wave-form substitution, etc.
- SIGTRAN (Signaling Transport)
- SCTP-Stream Control Transmission Protocol
- New options and new roles for Media Gateways replacing PBXs
- Expanded details on SBC-Session Border Controllers
- 50 point SIP security checklist
- More than 30 solutions to common problems and troubleshooting guide
- Improved section on QoS and RTCP-XR-MRB
- Indepth explanations of complex problems such as echo, crosstalk and asynchronous transcoding
- "Vo-eye-P" packet test
- Many other improvements, solutions, ideas and technology.

"The SIP Planning Guide course continues to improve in nearly every aspect of UC-Unified Communications, VoIP-SIP, OCS and IPT. This course is now even more vital to every organization as VoIP/SIP is new and new approaches to planning and managing are important to understand how. Coupled with the new improved solutions-answers, privacy/security section, troubleshooting guide and QoS primer, the SIP Planning Guide course are without peer from any source." noted Paul C. Daubitz - President - ATI-TeleManagement (http://www.ati-telemgt.com a professional management consultancy).

The SIP Planning Guide is available in Portland, Oregon. September 25-26th with additional dates and locations to be announced.

For more information contact Kim Jarrett at 888-251-5001 or kim@medak.com

About Medak
Headquartered in Portland, Oregon, Walt Medak & Associates, Inc. (WMA) was created in 1990 by Walt Medak with other former AT&T/Lucent/Avaya personnel specializing in service, maintenance and technical support of Avaya telecommunications equipment for interconnects, technicians and businesses. WMA offers Avaya telecommunications products, services and maintenance contracts as well as making available our RMATS (Remote Maintenance, Administration and Test Services) department to our own customer base. We also offer tech support to non-customers and interconnects. WMA has a complete stock of spares for each equipment type that we service. Additionally, we have functioning PBX switches to provide rapid disaster recovery (fire, flood, etc.). All our services are offered 24 hours a day, 7 days a week. WMA provides coverage throughout the continental U.S. from multiple locations. The company maintains a direct relationship in all major U.S. cities and has a nationwide network of associate dealers who distribute Avaya products and services; as such, we are able to quickly distribute Avaya products anywhere in the nation. Our national service center coordinates and manages all installations and provides RMATS to our customer base and associate dealers. For more information go to: http://www.medak.com/

72dpi_logo_with_disk.bmp

For those visually-inclined, go there for the animated tutorial:
response-point-intro.swf

One Size Fits Nearly All Businesses
Response Point from Microsoft is a really simple telephony solution for the less than 50 telephone user setting. Ideally, the number of users for their 4-8 analog business line solution is more like 20 phones. To put this in perspective, this size range covers the 5,255,810 out of the 5,885,784 or 89% of the U.S. businesses according to the U.S. Census Bureau (2004). This figure is only slightly larger according to Inc.com in 2006. In any event, Response Point fits nearly all SMB applications except for some high-volume small call center applications. The other key point is the simplicity of the system itself which consists of three components - a base unit about the size of a small book, SIP phones and optional gateway to the telephone company. If you want to use the corporate gateway in another city, you don't even need the PSTN gateway. There is software, of course, which provides for Outlook integration, phone features and really-easy customer administration. You could argue that this is not really anything different than what is available from a wide variety of existing vendors. However, the point is that this system is most likely to be available from your existing Microsoft solutions provider. This also means that the Microsoft channel partner can use their expertise in one more way to help the customer. That is, the customer gets one more product from one fewer and more integrated source. What has been found to be true in survey after survey, including those I have done for a lot of companies is that the number one thing customers want is - one stop shopping.

One-Stop Shopping
Distribution for Response Point is extensive. With more than 1,500 trained dealers and more on the way, Response Point is available in the U.S., Canada and with expansion elsewhere into English-speaking countries scheduled for 08/09 along with PTT and other localization. Response Point is even available at Costco. However, if you are not a DIY-do it yourself telephony or IT person, there are local dealers you can select from on the Microsoft Response Point website: http://www.microsoft.com/responsepoint/
One of the more innovative dealers is David Bainum of RiteTech (http://www.ritetech.net) who does a wiring "health check" prior to any installation. Most dealers recommend a complete network performance check due to the wide variety and age of computers, switches, routers, cabling and even electrical systems. David also recommends that customers install a backup power system to minimize outages and phone "reboots" due to outages along with POE-power over ethernet as Response Point phones and other SIP phones all require electrical power. Another Response Point dealer is Andrew Swingler who just launched http://www.ipphoneshack.com a one-stop portal for Response Point products and SIP trunking.

Pricing
Response Point is an exceptional simple system with a tiny hardware "footprint." The base unit hardware and telephone sets are manufactured by three key vendors D-Link (http://www.dlink.com), Aastra (http://www.aastra.com) and Quanta (http://www.syspine.com). Remember YMMV-you mileage may vary when it comes to TCO-total cost of ownership as customers may also need new cabling, switches, and other devices. While there are many sources for Response Point products, one example, not necessarily the lowest or the highest is the Syspine base unit price for four lines: $1999.00 and for eight lines: $2249.00 with the phones $159 each. Combination package deals run $2499 for base and four phones and $3599 for base and ten phones. $1270 for four lines and $1424 for eight lines. Aastra is the only system to date that offers a cordless system and D-Link does not support POE.

"Parts is Parts" - Key Components to the Response Point System
Here are the key components in Response Point. While, there are some differences between hardware manufacturers mentioned above, each of the three systems has the same Microsoft Response Point software.

Base Unit - Response Point IP PBX software running on Windows XP embedded steady-state storage for optimum hardware reliability - Built-in voicemail - VoIP gateway
SIP Phones - SIP endpoints (devices) connected to LAN switch - Response Point button enables one-touch voice commands - Auto discovery and configuration
Management & Client Software - HTML-based management console (on a PC in the LAN) for moves, adds, changes and managing system health - Client software on a client PC for Outlook contact integration, on-screen incoming call notifications and easy access to forwarding rules or personal preferences - Users can use as many features or add as many phones as they with NO additional software license fees

PSTN Gateway (Public Switched Telephone Network - local analog business telephone service)
- Only needed if access via there is no existing data bandwidth and SIP trunking gateway to PSTN
- 4 or 8 port base configuration with loop start signaling on analog business telephone lines (trunks are similar circuits but priced differently for use with PBX-Private Branch eXchange telephone switching systems) - Easy expandable in any line increment - No additional software or port licensing fees
Support for SIP Trunks
One of the key features in any new telephone solution is support for SIP trunks/trunking. Response Point in their latest SP1 release provides for support for SIP trunks. Like with any SIP trunk solution, Response Point is compatible with most SIP Trunk provider solutions. However, New Global Telecom (http://ngt.com), CBeyond (http://www.cbeyond.net/) and Junction Networks http://www.junctionnetworks.com/ all have been approved for their respective SIP trunk solutions. NGT also has a very comprehensive channel partner program that provides dealers with assistance in customer billing and other issues.

Bottom-Line
Response Point is the "new kid on the block" in a highly competitive marketplace. Maybe the world doesn't need another telephony solution but Response Point is more about office communications than just another piece of plastic on your desk, it enhances communications, simplifies installation and user activities, improves office productivity and time management and is competitively-priced. So don't believe me take a look for yourself at http://www.microsoft.com/responsepoint/

Reference: Detailed Feature Comparison
There is more information in this animated tutorial:

The following detailed section on comparisons of key features of Aastra, Syspine and D-Link. In addition, there is information on first-party, third-party call control, RCC-remote call control, CSTA-computer supported telephony application, SIP forking, SALT-Speech Application Language Tags specification, SMEX-Simple Messaging Exchange element and a number of other issues is only available to attendees at ITEXPO or to students in the SIP Planning Guide or OCS-101 online course.
This presentation is included in online/onsite courses SIP Planning Guide and for OCS-101 Office Communications Server per person (volume and site license discounts available). For more information, go to:
http://www.techtionary.com/sip/planning-guide/

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

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.

Microsoft's OIP-Open Interoperability Program Pries Open the PBX

As the alphabet soup gets more letters, the complexity of the enterprise OCS-PBX user also warms up. The animated tutorial explains the details which can be found at:
ocs-co-existence.swf

In other to create not just better Unified Communications applications but unified switching now being called software powered voice, Microsoft created the OIP-Open Interoperability Program to:
- Develop Industry-Class Telephony Infrastructure that work seamlessly with OCS and Exchange UM-Unified Messaging
- Develop many solutions and new ideas
- Provide a forum for customers with setup, support, and use
- Test to enterprise-level standards for audio quality, reliability, and scalability
- Provide a means for scalable qualification of vendors
For more on OIP, go here: http://technet.microsoft.com/ucoip

Also explained in the animated tutorial is the OIP program designed to provide PBX implementation/integration in the following configurations:
1 - Standalone via gateway
2 - Standalone via direct SIP
3 - Co-existence via dual forking
- Direct SIP + PBX is qualified against Microsoft Dual-forking specification
Co-existence via dual forking - dual OCS phone and PBX phone
4 - Co-existence via dual forking with RCC-Remote Call Control
- PBX supports Dual forking plus RCC-Remote Call Control and
- CSTA-Computer Supported Telephony Application
Here's how it might work using RCC-Remote Call Control a user uses their OC-Office Communicator client for Presence/IM and uses the OC softphone to control their existing PBX phone. For example, a user checks the presence for someone via Office Communicator and then clicking on that user to call - but then having their PBX desk-phone call the number (and use that device). Remote Call Control can also be deployed in conjunction with Dual-Forking using the "dual" calling of simultaneous or sequential ringing feature of "forking" to call the phone(s).

First Party Call Control
Traditional telephony POTS, SIP and OCS are designed to provide for first party or first person call control. Examples of first party call control are:
- Make/receive or accept/answer incoming or outgoing calls
- Make another call to another user
- Conference both the callers

Third Party Call Control
Third party or third person call control is where another element, endpoint, server, telephone or device is involved in the call. Third party call control may mean that the endpoints share call control with another device such as a PBX, ACD-Automatic Call Distributor, CO-Central Office Switch or other device. The third party device such as a server may direct, redirect (fork) or disconnect the call.

SIP Forking Explained
See the animated tutorial for details
Parallel Forking is where the proxy forwards copies of the request to multiple destinations simultaneously. This is simultaneous ringing where the phone rings your work, home, cell and any other phone at the same time. Forking is the process of processing multiple requests such as desktop, softphone and cell phones or finding the first-available or other call center rules and the proxy is called a Forking Proxy. A stateless SIP proxy will act as a simple forwarding process - ex: Ethernet switch. A stateful SIP proxy will review, route, fork, and modify the SIP header thus staying involved with the SIP dialog until BYE - ex: a Firewall.

Sequential Forking is where the proxy forwards copies of the request to one target at a time and waits for a final response (or failure) before moving to the next address. This is sequential ringing where your work phone rings three times, then forks to your cell phone for three rings, then your home phone for three times and then forks to voice mail.

RCC-Remote Call Control also known as third-party call control is provided by CSTA-Computer Supported Telephony Applications. CSTA was developed by the European Computer Manufacturers Association (ECMA) and subsequently was formally standardized by the ITU-T, incorporating the Switch-to-Computer Applications Interface (SCAI). CSTA uses, among other technologies, SALT-Speech Application Language Tags specification and its SMEX-Simple Messaging Exchange element, telephony call control capabilities in MSS-Microsoft Speech Server to allow a developer to create sophisticated telephony-based speech applications that can exploit both basic call control services such as ANI-Automatic Number Identification (caller ID) and DNIS-Dialed Number Identification Service (800), using the included basic call controls, or extended call control services, to create custom call controls. CSTA uses ASN.1-Abstract Syntax Notation One - a notation system for describing data structures. ASN.1 while like programming language is in fact, not a programming language. ASN.1 is a flexible notation that allows programmers to define a variety of data types. ASN.1 is a set of encoding rules used to transform data into a standard format that can be decoded on any system that has a decoder based on the same set of rules.

SALT adds voice commands to web applications. SALT consists of a series of software applications meta-tags that are captured, interpreted and responds to audio-voice and DTMF-Dual Tone Multi-Frequency (touch tone) input. 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. Many of the features of SALT include:

- Multi-mode browser-driven clients - PDA-Personal Digital Assistants, etc.
- Authoring Tools - IVR-Integrated Voice Response, grammar, dialogs, commands
- Speech-driven WEB pages with voice/data databases
- Interface to analog-digital telephones within SIP/VOIP servers and integrated voice recognition
- XML services with other business services such as SOAP to OLAP database connections
In the animated tutorial is an explanation of SALT in the OSI model.

In the animated tutorial are some of the basic SALT commands such as: - Prompt - Used to play audio prompts to direct a user to enter input - Grammar - Used to identify how audio sentences will be analyzed - Record - Used to trigger saving voice input

The concept of software powered voice is very compelling as you make voice just another XML service.

This presentation is included in online/onsite courses SIP Planning Guide and for OCS-101 Office Communications Server per person (volume and site license discounts available). For more information, go to:
http://www.techtionary.com/sip/planning-guide/

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

See animation for visual explanation of SIP DOS-Denial Of Service attack.
sip-security-asn-overflow.swf
This is an example how an attack could occur. The attacker can alter the encoding lengths causing a buffer overflow-overrun at the endpoint. According to Cisco, "The attackers can try to use PER encoding coupled with the ASN.1 representation to encode excessive recursive fields and lead to huge processing and memory overhead at the endpoint."

While it is beyond the scope of this tutorial to explore all types of VoIP security violations and attacks, attackers can try to compromise H.225 protocol implementations. That is, since H.225 messages are ASN.1 PER-Packed Encoding Rules encoded (or compact binary encoding on limited bandwidth networks), the attacker can alter the encoding lengths causing a buffer overflow-overrun at the endpoint. ASN.1-Abstract Syntax Notation One is a notation system for describing data structures. ASN.1 while like programming language is in fact, not a programming language. ASN.1 is a flexible notation that allows programmers to define a variety of data types. ASN.1 is a set of encoding rules used to transform data into a standard format that can be decoded on any system that has a decoder based on the same set of rules.

This presentation is also included in online/onsite courses SIP Planning Guide and for OCS-101 Office Communications Server per person (volume and site license discounts available). For more information, go to:
http://www.techtionary.com/sip/planning-guide/

Discounts are also available to members of the SIP Forum and MS Partners. 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.

SIP-Trunk Planning & Training Guide 2.5

More than 100 New Concepts for Comprehensive Planning, Implementation, Security, Troubleshooting Guide for UC-Unified Communications, SIP, OCS-Office Communications Server Networks

BOULDER - August 12 - TECHtionary.com today announced "SIP Planning Guide 2.5," a significant expansion of what's been called the "quintessential" planning guide for SIP-VoIP, UC-Unified Communications, Microsoft OCS-Office Communications Server & Response Point and IPT-Internet Protocol Telephony networks. SIP Planning Guide 2.5 provides the means for customers who are planning, implementing and expanding their communications networks to "get SIP smart" with guidance, ideas and tools. With more than one hundred new concepts for review, SIP Planning Guide 2.5 is vital to anyone doing UC/VoIP/SIP/IPT for network solutions and premise-based implementations.

Some of the key highlights in SIP Planning Guide 2.5:
- VLPS-Virtual Private LAN Service with MPLS
- Privacy-Security Lifecycle Management - Authors - Actualizers - Auditors - Analyzers
- PLC-Packet Loss Concealment - zero Insertion, wave-form substitution, etc.
- SIGTRAN (Signaling Transport)
- SCTP-Stream Control Transmission Protocol
- New options and new roles for Media Gateways replacing PBXs
- Expanded details on SBC-Session Border Controllers
- 50 point security checklist
- More than 30 solutions to common problems and troubleshooting guide
- Improved section on QoS and RTCP-XR-MRB
- Indepth explanations of complex problems such as echo, crosstalk and asynchronous transcoding
- "Vo-eye-P" packet test
- Many other improvements, solutions, ideas and technology.

"SIP Planning Guide 2.5 continues to improve in nearly every aspect of UC-Unified Communications, VoIP-SIP, OCS and IPT. This course is now even more vital to every organization as VoIP/SIP is new and new approaches to planning and managing are important to understand how. Coupled with the new improved solutions-answers, privacy/security section, troubleshooting guide and QoS primer, SIP Planning Guide 2.0c courses are without peer from any source." noted Paul C. Daubitz - President - ATI-TeleManagement (http://www.ati-telemgt.com a professional management consultancy).

"SIP Planning Guide 2.5 takes our SIP/OCS courses to the critically important next level," noted Tom Cross TECHtionary CEO. "SIP is a complex process because SIP is not just a single location service but one that encompasses nearly all aspects of business communications and computing," Cross added.

SIP Planning Guide 2.5 is available in the onsite and online courses. The online version is $199 per person or four (4) for $499. Add-On OCS Module - there is a special OCS-101 Office Communications Server module for $99 per user or less with discounts. Special SIP Forum Member and Microsoft Partner price is $99 per user.

For a complete detailed course outline go to: http://www.techtionary.com/sip/planning-guide/

Podcast Series on Selling SIP Systems & Solutions - "You Can't Sell What You Don't Know"
By Thomas B. Cross CSSP - CEO TECHtionary.com
I just taught a class to new-hires to the SIP industry and often get asked by nearly everyone "what is SIP?" One answer that I increasingly give is "in the old days, phones were dumb like POTS and networks were smart like the PSTN. In the new internet world, phones are smart and networks are dumb." Check the animation here to see what I am talking about.
dumbphones-smartnetworks.swf
If you prefer a more technical tutorial, here goes. With SIP Trunking, the IP media stream coming from within the enterprise stays as an IP media stream and passes to anywhere within the enterprise or across the boundary of the enterprise to another enterprise via IP. This reduces the need for local telephone systems using instead hardware media gateways at the enterprise edge and carrier edge (often referred to as the PSTN) producing considerable savings. In addition, considerable savings can also be found in eliminating expensive telephone desksets by using intelligent softphones.
In addition, if you want to sell new SIP solutions, you need to know a lot more. At a recent conference, the panelists on the session Selling SIP Trunking 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 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 benefits when users are not on the telephone with the bandwidth automatically or "dynamically" available for 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 dozens of 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 Planning Guide 2.5 available in the onsite and online courses. Discounts are also available to members of the SIP Forum and MS Partners for $99 per student. 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