Want a really good shot at winning a free iPod Touch?  If you are headed to VoiceCon in Orlando later this month, you should definitely read on...

AudioCodes has joined forces with ScanSource Communications at this upcoming VoiceCon in Orlando and as part of the exhibition, we are hosting a "Solutions Theater and Pavilion" in our expanded booth #931.  We are thrilled to have pulled together 14 industry leaders that will deliver a series of presentations that focus on SIP-based applications that help end users and VARs deal with this difficult economy.  

Now for the really cool part: After each presentation, AudioCodes and ScanSource Communications is giving away an iPod Touch to one of the lucky audience members.  (Rules for the drawing will be posted in the booth)

Time	Presenter	Topic
4:00 PM	BandTel Joel Maloff Senior Vice President of Sales and Marketing	SIP Trunking
4:30 PM	CTI2 Erez Marom	Unified Communications

Time	Presenter	Topic
1:30 PM	AudioCodes Alan Percy Director, Market Development	IP Communications - An Opportunity in a Down Economy?
2:00 PM	Avaya Bruce Mazza	Branch Office Solutions
2:30 PM	The VIA Group Jeff Stillings	Microsoft Office Communicator 2007
3:00 PM	Genesys Charles Lee      Sr. Product Marketing Manager	Empowering enterprise-wide customer service
3:30 PM	Atlantic Communications Michael Light	Hosted Solutions (Cosmocom)
4:00 PM	Digium Bill Miller VP Product Management	Asterisk Open Source Solutions

Time	Presenter	Topic
1:30 PM	Brian Cuppett ScanSource Comm.	ScanSource Communications
2:00 PM	Strategic Products and Services (SPS) Mike Taylor, CTO	Avaya Branch Office Solutions
2:30 PM	Interactive Intelligence Rick Q. Chin Manager, Solutions Marketing	Improving the Customer Experience with CIC
3:00 PM	EUS Networks Robert Campozano CEO	Asterisk Solutions on Mediant 1000 OSN
3:30 PM	Sagem-Interstar TBA	Enterprise Fax Solutions
4:00 PM	Enabling Technology Steve Bruno	Deploying Microsoft Office Communicator

See you in Orlando!

  
 The week of February 2nd takes me to ITExpo in Miami Beach.  A much needed break after the big snowstorm that hit the Northeast and a chance to meet face-to-face with others in the industry.

After spending a couple years of experiencing the social networking evolution, connecting with with Linked-In, Facebook and other social networking resources I've started to re-think my understanding of industry conferences.  One might think that with all the virtual and on-line tools we have (teleconferencing, Webex, Facebook, etc.), industry conferences might fade into obsolescence.  However, I would argue the opposite might be the case.  Certainly the tools have helped me keep connected with my industry contacts (and in many cases, get even closer than before) but there is no replacement for a face-to-face visit, a handshake, or a drink together.  In fact, I feel the need is even stronger to get some much needed face-time with these people.  Why?  Because with social networking tools, access to each other is easier and everybody is competing for my time and others.   In the face-to-face world, I can cut through all noise and get quality time and converse with those that I really need.

So, what do I have planned for this next week?   I've put together three great sessions that I hope my readers will take the opportunity to participate in while at ITExpo:

"SIP Interoperability: The Ultimate Myth?"
MON 2/2 -- 12:00-12:45pm
TRACK: DEV-01 -Developer
ROOM #: B212
This is a discussion on the real-world of SIP interoperability.  I spent quite a bit of time with the director of our interoperability lab, discussing the challenges that he sees when trying to complete interoperability between two SIP-based devices.  I've brought alone some thoughts and ideas on how resellers, OEMs, integrators and others can solve tough interoperability issues in their solutions.  No product pitches here, just solid advise from real industry insiders.

"HD - What's the noise and are we ready?"
TUE 2/3 -- 12:00-1:00pm
TRACK: L-01 -Special Luncheon Panel
ROOM #: B216
Back a few months ago, we launched our HD VoIP strategy and with that found that few people understand what HD really is and the impact it will make on our industry.  Working with Rich and the staff at TMC, we're taking this conversation to a much wider scope and bringing together a number of experts on HD to discuss how we can make our joint dream become reality.
 
"New SIP Trunking Announcements"
MON 2/2 -- 2:00-2:45pm
TRACK: RES-03 -Reseller Day
ROOM #: B116/B117
In response to the economic melt-down, we sat down and came up with a short list of opportunities for our industry could leverage to not only get through the down-turn, but better position themselves for the recovery.  One of these opportunities is SIP Trunking for existing SMBs that have existing TDM equipment and need to save operational costs without large capital expenditures.  I'll introduce our "SIP Trunking As You Are" strategy and provide the tools attendees will need to leverage this opportunity.

And, as always, I'll be racing from one meeting to another during the event.  If you'd like to get on my calendar, please drop me a note or stop by our booth #616 during the event.

If you remember a few weeks back, I mentioned that we were seeing a growing opportunity in the market for SIP Trunking providers connecting their services to the very large installed base of TDM PBXs, KSUs, Contact Centers, etc.   In a number of conversations I've had with both our VARs and with end-customers, a pattern has clearly evolved in the market that is a perfect fit between SIP technology and business needs.

The Business Problem

With the stress on businesses today, finding ways to save on communications costs is a major concern. Replacing aging PBX equipment would save some operating costs, but the CFO is adverse to large capital expenditures, trying to conserve cash and keep their lines of credit available for unforeseen troubles ahead.   In addition to the financial challenges, a change-out in PBX is a very complex and disruptive process, needing time to find the right application and do proper evaluations.   Is there is a solution that can save operating costs today and meets the long-range needs for the organization in mind?

The Technical Problem

The TDM PBX is still in perfectly fine working order, but is clearly a dinosaur with limited SIP capabilities without being replaced.  Because it is older and TDM, it cannot directly connect to the new SIP Trunking services without major upgrades.  

The Solution: SIP Trunking As You Are

In a nut-shell, this strategy takes customers that are using expensive dedicated voice T1/E1 or analog trunking from the local telco and replaces them with a SIP connection to a SIP Trunking service provider as shown below:Today we participated in a joint announcement with Broadvox, one of the leading and most aggressive Internet Telephony Service Providers (ITSPs) that offers a very comprehensive SIP Trunking offering.  In our discussions with David Byrd, VP of Sales and Marketing for Broadvox noted: "During this challenging economic period, businesses are looking for ways to save on communications costs with as little capital expenditure as possible. "With Broadvox SIP Trunking and the AudioCodes Mediant 1000, businesses can start saving today without having to replace their PBX dramatically improving their ROI."

We are holding a special web-based seminar to discuss this on Wednesday, January 14th at 2 PM ET.  To register for the event, visit: 
http://www.audiocodes.com/events/-sip-trunking-as-you-are-reduce-costs-add-flexibility-keep-your-tdm-pbx-and-legacy-cpe

Some great questions were raised during the most recent webinar titled "IP Communications - an opportunity in a down economy" and I thought sharing the entire list would be valuable.  So here it goes:

Q: What needs to be done to the typical PBX to use SIP Trunking?
A: Most PBXs, Key Systems and other TDM equipment connect to the public network using either analog or T1 trunks here in North America.  In EMEA and other areas, you'll also see E1 and ISDN BRI trunking.  For SIP trunks to connect these legacy systems, a SIP media gateway with matching TDM interfaces is required.

Q: What kind of interoperability issues are there between SIP trunks and IP-PBXs?
A: Right now there are plenty of interoperability issues and while there is some progress on standardization, is seems that it will be a while before you can just "plug and play".  The issue is the looseness of the SIP standard and investment many vendors have already made in their networks.  The end result is that today some either software or hardware device must do the IP-to-IP mediation between the two different formats, converting both signaling and media as required.  I put together a paper on the topic last year that you may find helpful.

Q: Do you see more opportunities in the residential or in the business segment during this recession?
A: It seems that today business are the most eager to cut operating costs.  Residential customers seem to be frozen in their tracks or just cutting the cord to their land-lines.

Q: What new products does AudioCodes have for hosted PBX providers today?
A: We have CPE gateways for the customer premise, large media gateways that connect the hosted application to the PSTN, session border controllers that secure the connection between the service provider and CPE, and media servers required for conferencing, announcements and other applications.  Check out our new web site at www.audiocodes.com and use the application navigation page for more ideas.

Q: Does SIP Trunking replace the existing T1 completely?
A: In many application - Yes.  The legacy TDM Voice T1 is completely removed.

Q: What changes are needed in the TDM PBX to use SIP Trunking?
A: In many cases - none.  The media gateway would be configured to emulate the legacy TDM T1/E1/BRI/Analog trunk allowing the PBX to continue to operate as-is.  In a rare few cases, a few parameter settings within the PBX would need to be changed.  However, there is no need to add any cards or spend any more money on the PBX.

Q: I keep hearing that SIP carrier trunks are not available in all markets? Can you please comment on that .?
A: Some of the SIP Trunking service providers do have a regional focus.  This allows them to better address and service their selected regions.  Others are more global.  When you call a SIP Trunking service provider, make sure you have your service locations in hand and they can tell you if they can service those sites.

Q: On slide 16, what is "OSN"?
A: Open Solutions Network - this is the embedded application server that is part of our partner network program, allowing them to run their applications within the gateway.  You can think of this as an Intel server sand-box within the gateway.  This eliminates a separate server and allows for one-box appliances.

Q: How is VoIP security such as Session Border Controller functions addressed by AudioCodes?
A: How a Session Border Controller (SBC) works is a very complex topic and a paper on our web site explains it in detail, but in a nut-shell it uses a Back-to-Back User Agent to terminate the SIP sessions, allowing the examination of the request and comparing it against a number of security rules.  Once validated, the request is re-issued on the other side and sent along the way.  A similar process may also occur with the media streams.

Q: Can the Mediant 2000 support your SBC module?
A: The Mediant 1000 has an SBC module, but today the Mediant 2000 does not have an SBC module.  An external SBC like our nCite 1000 may fit the application/

Feel free to use the Comment feature to post more questions and keep the dialog going!

Thanks everyone that participated in yesterday's webinar.  It was interesting to see the diverse range of participants and responses from the session.

If you missed it, here is some background on the topic:

This most recent economic downturn has far reaching impact on a wide range businesses and industries. In this climate, corporate executives are examining every capital expenditure to ensure there is a quick return on investment and that the expenditures are in line with the strategic goals of the business. This has put many large communications infrastructure capital improvement projects on hold while alternatives are examined more closely. However, even in a down economy there are some very clear opportunities for businesses to reduce current operating costs without large capital expenditures. This session will examine a number of these opportunities and examine real customer case studies of projects that demonstrate of how operating expenses can be reduced, while staying on track with strategic goals. Join us to find out how you can identify and leverage these opportunities, even in a down economy. 

During the event, we discussed a number of areas where opportunities exist to help your business or customers save operating costs, without large capital expenditures:

• SIP Trunking for both TDM PBXs and IP-PBXs
• Software Migration Solution
• Open Source
• Software as a Service
• Enterprise Networking

An on-demand recording of the event is available by clicking here.

When we receive a copy of your questions posed during the session I'll post another entry with answers (to the best of my abilities).

Also, if you have any suggestions or further questions, please leave a comment here.  This is a two way and open conversation!

This last few months we've started to see growing opportunities with SIP Trunking partners, helping them with media gateways to connect their services with end-customers that want to retain their legacy TDM PBXs.  



There are a number of reasons for this interest:

Cost Reduction - SIP Trunking allows SMBs to reduce their local and long distance charges and eliminate the need for separate T1 telephone circuits.  With SIP Trunking, all their voice and data traffic share the same physical last mile connection.

Saving the PBX - The vast majority of today's installed base is still using TDM PBXs.  Many SMBs and enterprises upgraded their TDM PBXs back in 1999, preparing for Y2k.  Many of these are still working perfectly and have years of useful life remaining.  Why toss out a perfectly good business tool, especially with the current economic situation?

Simplying The Process - instead of trying to decide on a complete communications infrastructure upgrade and try decide on a new IP-PBX, just upgrade the part they need now (cost savings).  The typical IP-PBX decision process takes close to a year, issuing RFIs, evaluations, getting buy in from all the departments.   Start saving money now!

Security - until all the issues with SIP security are fully addressed, this architecture is the most secure means to keep hackers out of your network.

Right Place at the Right Time - To date, most of the noise at the industry trade shows has been about SIP Trunking with IP-PBX, which fills the rooms and creates buzz.  However, what about the millions of TDM PBXs out there?  It seems to me that this is the bigger market that can be addressed now.

Stay tuned as this discussion continues....

As part of our continuing series on applications based on SIP, I've pulled together Mike Wilkinson of Broadsoft and Scott Firth of IBM to discuss the challenges and solutions that small communications operators can leverage SIP and VoIP to expand their business.   The objective of the event is to educate the smaller and rural telecos, internet service providers, cable companies on the opportunities and solutions available to add voice to their existing IP networks.

The story behind the story starts on my front lawn with the owner of a local fixed wireless ISP we have here in western NY.  His network uses fixed wireless repeaters around northern Allegany County, NY to service hundreds of homes, farms and businesses with broadband that the local ILEC won't service with DSL or other wired broadband technologies.  He was describing his challenges with building out the network and desire to add services to increase revenue.   The rest of the story will be covered in today's event.

Small Operator - Are you being left behind?

Wednesday, November 19, 2008 2:00pm ET / 11:00am PT

Small and mid-tier operators of cable systems, rural telcos, internet service providers, and regional wireless providers are constantly challenged with ways to improve customer retention and increase revenue.

VoIP and other bundled offerings are widely recognized as a means to achieve these goals. Market analyst AMI Partners Inc. agrees, noting in a forecast that SMB hosted VoIP spending will surpass $1.56 billion by 2010. However, the cost, technical complexity and financial risks involved with building a VoIP solution from scratch have been outside the means of many operators and as a result, kept them on the sidelines.

This situation has allowed the unbundled VoIP operators to draw away customers and revenue while draining resources from the operator's network.

The event will be available for on-demand view, just follow the same link to access.

This week here at AudioCodes has been very busy.  You may have seen the post on Monday, announcing our HD VoIP strategy  which will dramatically improve the clarity and quality of voice communications.

Yesterday, we announced our new line of IP Phones, all of which will support HD VoIP.  As far as I can tell, we will be the only manufacturer that will have a 100% HD VoIP capable phones (even the low-cost entry model).

The market analysts seem to agree:

• The 310HD IP Phone is positioned as an entry level IP-Phone and includes a basic display and user interface. 
• The 320HD Premium model includes a large Monochrome LCD screen. 
• The 350HD Executive model has a large Color LCD. All models support HD VoIP.

The phones will include many important features for a range of applications, including:

• Support for popular wideband coders such as G.722, G.722.2 (WB-AMR), G.729.1 and G.711.1.
• Power over Ethernet is optional in all models.

The products will be available for testing and evaluation beginning in February 2009.

To more information on the devices or HD VoIP, click here

Okay, it's the 21st century and there are many new innovations and technologies that make our lives a whole lot easier, efficient or entertaining.  Think back about life before cell phones.  Remember pagers and calling cards?  How about the changes in TV?  With super clear picture and surround sound, HD TV makes you feel like you are at the game.  Things sure have changed for the better over the last twenty years.

Well, with one big exception - the voice quality on your telephone.

You see, the current Public Switch Telephone Network is built on technology invented in the late 50's based on digital sampling of your voice using Pulse Code Modulation (PCM) sampling.  Back then, it was groundbreaking improvement in reliability and clarity.  But to use the infrastructure and cabling efficiently, they had to make some choices about how much of your voice to collect and transmit.  The choice was a cost/benefit decision that came up with a 3.4 kHz bandwidth that created a "sound barrier", limiting the fidelity of your voice ever since then.  

Why is this important?  The 3.4 kHz bandwidth limitation in the PSTN is universal, allowing carriers to interoperate and pass voice from one to another.  It's also the ultimate commodity  - "one size fits all" in communications.  No matter whether you use one of the Bell companies or a smaller competitive carrier, everything sounds the same.  The result?  Price wars and customer churn to chase the ever cheaper commodity service.

At least until now.

With VoIP and SIP working together, we finally have the tools at our disposal to dramatically improve the quality of voice communications and break through the "sound barrier" with VoIP that uses higher sampling rates and new voice coding algorithms.

We here at AudioCodes are quite pleased to announce our HD VoIP strategy that we feel will play a critical role in migrating both the wireline and wireless communications infrastructure away from the limitations of the PSTN and into the future of High Definition Voice over IP (HD VoIP).  HD VoIP will allow carriers to differentiate their services with much higher quality voice calls and create affinity amongth their customer base.  Enterprises will be able to improve efficiency and reinforce their branding with high-fidelity customer contact.

Want to learn more?  See our dedicated landing page at:  www.audiocodes.com/hdvoip

Or attend the live webinar that I am hosting on Tuesday, November 18th at 2:00 PM EST.  Click here to visit the Webinar Registration Page

I'm here at Astricon in sunny Glendale, AZ with the diverse Asterisk community talking about using SIP to improve the reliability and scalability of Asterisk.  

We had a full room this morning during our session sponsored by both ScanSource and AudioCodes, discussing different architecture tricks to improve reliability of Asterisk solutions and spread out the risk with the goal of avoiding a "career affecting" event.

A quick summary of our recommendations:

1. Use redundant and commercial grade servers to avoid simple power supply and fan failures from taking out the whole system.  
2. Use a distributed SIP architecture to separate the Asterisk server, media gateway and phone devices.
3. Implement a load-balancing scheme to spread the traffic over both Asterisk servers and dual media gateways.
4. Use Dundi to keep the dual Asterisk servers synchronized and allow them to cover for each other.
5. ...and many other helpful tips....

We had a number of interesting questions about how Asterisk handles routing the RTP streams around the application server, enabling mid-call fail-over in the case of an Asterisk crash.

For more details and/or a copy of the presentation, visit the AudioCodes micro-site on the ScanSource web site at: www.scansourcecommunications.com/audiocodes