Earlier this month I was happy to meet with Jeff Heibert, CEO of ROI Networks to LA for ITExpo for a discussion on a case study customer of theirs using Skype Connect in an Avaya environment using the AudioCodes Mediant 1000 E-SBC.  The E-SBC plays a key role to integrate the two systems and provide security to protect the Avaya SIP Enablement Server (SES) from attack via the public internet.  After some nudging, I got Jeff to sit with Erin to do a video interview of what we think is a fascinating case study.


Click on the thumbnail to view the video.

A key attribute of this case study is the end customer was able to leverage Skype Connect without expensive and disruptive upgrades to their Avaya Communications Manager.

A written copy of the case study is also available.

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• Security - knowing that your business is protected from malicious attack by outsiders that might have an axe to grind or trying to steal sensitive information.  
• Survivability - making sure you have a disaster recovery plan to keep the business operating when things do go wrong.
• Choice - having the freedom to make changes to and select your service provider based on price, service, quality or new capabilities.

Until recently, if you were in the process of deploying Microsoft's Communications Server it was pretty difficult to use SIP Trunking and still maintain security, survivability and choice.   Securing SIP Trunks requires a Session Border Controller at the enterprise from one vendor, while survivability required separate media gateways and back-up PSTN telephone lines.  The number of certified SIP Trunking vendors that have gone through the arduous certification program is fairly small and as a result, severely limits the choices available to most businesses.



With today's announcement of UcSIPT from AudioCodes, businesses deploying Microsoft Communications Server can leverage an AudioCodes E-SBC - one single device that has the ability to:

• Secure SIP Trunks - with an embedded Enterprise Session Border Controller (E-SBC), protecting the business against common SIP-based network attacks coming from the Wide Area Network that traditional firewalls are unable to mitigate.
• Provide Interoperability with SIP Trunks - via a back-to-back user agent with transcoding, virtually any SIP Trunking service provider can be used with Communications Server.
• Deliver Survivability - with an optional integrated media gateway modules, traditional PSTN lines can be maintained and used as back-up circuits in cases where there is a failure in the SIP Trunk connectivity or service.

As an example of a typical deployment, Joe LeNoach, IT Manager in Alutiiq offers the following assessment of his experience in using the Mediant 1000 MSBG to interface his Office Communications Server system with a SIP Trunking service provider:  "We have recently decided to use the AudioCodes Mediant 1000 for connecting our network to the Verizon SIP Trunking service," said Joe. "We were looking for a solution to securely connect the Microsoft Office Communications Servers installed in our branches using a cost-effective SIP trunking service. AudioCodes Mediant 1000 provided the required mediation services to implement this solution quickly and cost-effectively."

Jeff Kahn. Chief Strategic Officer and Nimrod Borovsky, VP of Marketing at AudioCodes discuss the UcSIPT strategy and solution set in this video.

It looks like businesses are one step closer to having the choice of SIP Trunking service provider, security and survivability without cobbling together a number of different appliances from different manufactures.  

For more information on UcSIPT, visit: http://www.audiocodes.com/ocs








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While on the road this spring I had a number of very active conversations with our partners and customers about the delivery mechanisms, services and bundling of SIP Trunking here in the US.  It seem there are some patterns that I thought would be useful to share.

First I'm going to assume that you know that SIP trunking is a replacement for legacy TDM trunking lines that connect various size enterprises to the public network.  SIP trunks offer similar services, but instead of using dedicated TDM T1/E1 or analog telephone lines, the voice traffic is transported over IP-based data circuits.  SIP is used as the signaling protocol, controlling the start and stop of each voice conversation, associated caller ID and other enhanced services.

SIP trunks are not all created equal though - there are some very significant differences in the way they are sold and the services they support. I could spend months going into all the various technical and business model differences between the vendors, but today I'm just going to concentrate on the different ways they are delivered to the customer.

Tightly Bundled SIP Trunks
These typically are tightly tied to data services that would come from the service provider.  In this case, the service provider requires that you use their data infrastructure to carry the SIP trunks. They arrange for the last mile circuit, provide all the equipment and provide the services with one bill. The logic the service provider uses for this offering is that unless they can control the data infrastructure end-to-end, they can't guarantee the voice quality.  The biggest benefit of the bundled services is that it gives the customer "one throat to choke" if there are issues with the service or quality of calls.  

However, I've heard some push-back from customers on this "take it or leave it" business model, requiring that they buy both the broadband and voice services together.  In some cases the cost reduction doesn't justify the complexity and risk.  What if the enterprise already has an established relationship with a broadband provider and is under contract?  Do they need to pay to break the contract?  What about the risk of potential disruption while migrating both the voice and data services to a new service provider?  Can the service provider deliver both voice and data services to all my offices?

In the end, it seems tightly bundled SIP trunks are best suited to greenfield deployments within the service provider's area of coverage.

BYOBB SIP Trunks
The opposite of the above separates the SIP trunks from the broadband delivery to the customers site.  In this case, SIP trunks are a service that the customer uses on their existing or upgraded broadband facilities.  Thus the term Bring Your Own BroadBand (or BYOBB) was born.

The advantages of this service offering is that it can be offered virtually everywhere where there is sufficient broadband and it can be layered on top of existing Internet services. This allows an enterprise to partially or fully migrate to SIP trunking without disrupting their current data services.  For some enterprises, they have already done extensive upgrades to their data infrastructure and SIP trunking is just another application that was already budgeted for well in advance.  The ability to choose multiple SIP trunking services separately from the broadband is a powerful tool when negotiating on a service contract, especially when looking for local numbers outside the US.

Many of the BYOBB SIP trunking service providers let you choose the equipment at your premise.  From WAN access, the router, security solution and the media gateway.

On the downside, I've heard some debate about the validity of the methods used to test the existing broadband circuits and how to prove compliance with service level agreement terms, especially with voice quality.  Then there is a long-term problem of the broadband carrier managing the traffic inside their network.   If it works today, will it work tomorrow as they add new customers? It's important to choose a service provider that supports built-in quality monitoring capabilities, including RTCP-XR which reports real-time R factor scores on voice quality.

Can You Have your Cake and Eat it too?
There are a number SIP trunking service providers that try to ride the fence and will offer you either a bundled service or unbundled, based on your individual situation.  However, it sounds like you will get pressured hard to take the bundled service to control quality and "maximize value".  The bundled services are frequently wrapped up in one-size-fits-all packages that are a lot easier for them to sell, install and service.  The only question is:  Do you fit the one size they are offering?

What's Right for You?
Well, it really depends on a few factors:

1. How much do you value having "one throat to choke"?
2. How much bandwidth do you currently have and is it voice-ready?  If you already have a good broadband provider, use it!  If not, maybe a bundle would get both upgraded at once for a good price.
3. Are you under contract with either a voice or data service provider?  Does it make sense to break either or both the contracts?  Work the numbers - then decide.
4. How much control do you want over the equipment and services?  If you could care less, just go with a bundle.  The more you know, the more control over the services and equipment you will invariably want - go with unbundled services.
5. Are you planning a slow migration with a few circuits to start or are you going to cut over all at once?  Complex and gradual cut-overs need more control.

Hopefully this background on the range of SIP trunking offerings will help you with your adoption.  Make sure you ask the right questions and consider your individual situation before signing anything!
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AudioCodes has been working closely with Avaya for a number of years and as part of our exhibition at VoiceCon Orlando next week, I've invited Bruce Mazza, Director of Branch Solutions at Avaya to present during our AudioCodes / ScanSource Solutions Theater.  Avaya's Branch Office solution leverages SIP to connect remote branch offices to regional offices, which reduces operating costs for banks, insurance, retail and government customers that frequently have dispersed remote offices.

AP :Where would someone learn more about the Avaya Branch Office solution?    

BM: We'll have information at both the Avaya and AudioCodes booths with live demonstrations that can be seen by attendees.

Bruce's presentation is scheduled for Tuesday, March 31st at 2:00 PM EDT in the AudioCodes / ScanSource Solutions Theater within booth #931.

Did I mention that by attending you can win one of 14 iPod Touch PM3 players that we are giving away?

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 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.

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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! Continue Reading...

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. Continue Reading...

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....
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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. Continue Reading...

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

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