The SIP Invite

It's been a while since my last post on my Verizon FiOS and Tivo experience and a few things have changed, so I thought it would be time for an update.

Tivo Upgrade - first and most importantly, sometime over the last couple months, the folks at Tivo have finally figured out a software solution to the picture "pixelation" that was occurring with Verizon FiOS.  Sometime ago, my Tivo automatically updated to the latest 11.d version software and to test the upgrade, pulled all the attenuators out of the line.  Results: it works!

With the picture quality issue now behind me, it was time to finally buy that second Tivo for the TV in our master bedroom.  It arrived a couple weeks ago and installation was much easier (now that I know the drill).  

By the way - one very important installation detail:  Tivo HD is shipped from the factory with an old version of software (really old), so you need to give it a day after you complete the initial configuration to perform the automatic software upgrade before worrying about the picture quality with Verizon FiOS.  Once the upgrade is completed, the picture quality should be corrected.  

I really love having two Tivos in the house now - the networking feature allows you to record on one Tivo and play back on the other.  Sounds silly, but it has already cured a couple domestic disputes in my household.  The choice between watching "Dancing with the Stars" or "How It's Made" had required moving to different rooms/TVs.  Now it doesn't matter which Tivo records the show, you can watch it on either one!

Closing Thoughts:
After all the original installation headaches, it seems to me that Tivo HD and Verizon FiOS are now working perfectly together.  Great user interface, built-in NetFlix client, fantastic picture quality - I finally have my peanut butter and chocolate!

Two weeks ago, I spent four days at CableLabs - talking about and seeing the new developments in the cable television business.  Truly fascinating stuff with new DVRs, on-demand programming and even a 3D TV demonstration.  However, based on what I seeing at home, I suspect someone is about to "move their cheese".

If you have been following my now year-long adoption of Verizon FiOS, you know that we've been using a TiVo-HD with FiOS to record and watch television. We subscribe to the FiOS triple-play bundle with HDTV, 20Mbps Internet and two work telephone lines.  As in most households, my teen-age kids have been the primary users of the television all summer and I've been watching their behavior on how they use the television, what they watch and how much time they spend watching.

They spend hours in front of the TV, but they don't watch "TV".  No, they don't play video games - it's NetFlix over the Internet that has become their primary source of visual entertainment.  You see, the TiVo-HD includes an optional NetFlix on-demand client that allows the kids to choose programs from the vast NetFlix library and then watch them instantly on our TiVo via the Internet.  It really is amazing - no waiting for the DVD anymore - just click and watch.  The kids have even put the NetFlix mobile client on their iPod Touches, allowing them to browse the NetFlix library from the couch.  

So what does this mean to the Cable Industry?  Think about it - here is a generation that who's world is a broadband connected and able to access virtually any/all information almost instantly.  And most of that information comes from other sources, not the cable operator.  Based on their usage, the cable operator is just a supplier of broadband Internet.

This really means that when the millennials grow up and start controlling household spending, things will change dramatically.  Our generation values the TV programming, Telephone, then Internet in that order.  I suspect they will reverse it, with their highest value being the Internet and then TV and Telephone being almost irrelevant.

Where does this take both the cable companies and legacy wireline operators?   Their focus will soon need to shift to reliable and very high-speed Internet access being the "high value and primary product", because the rest is quickly going the way of the Dodo bird.

HD VoIP is one of key strategic topics being discussed here at the CableLabs Summer Conference 2009 in Keystone, CO.  

My week started with a discussion of the future of voice services on cable networks and a live demonstration during the CableLabs Innovation Showcase, highlighting  a number of audio clips that showed the difference between the existing narrow-band voice quality and what the same clips sound like with HD VoIP.  During my presentation, I challenged the cable operators in attendance that unless they do something to differentiate their voice services, that wireline telephony will soon fade into oblivion, taking away an important revenue stream from their services, turning the "triple play" into a "double play".  I was joined by Larry Clarkson, Product Marketing for the AudioCodes System Group who helped me present a number of architecture diagrams showing how HD will be integrated into the current and future network infrastructure.  We wrapped up with a demo featuring a live transcoding demonstration of a wide-band call between a Nokia E71 mobile, a DECT phone and AudioCodes 320HD phones.

Tuesday's main theater session is "High Definition Voice" featuring a panel of experts including Jean-Francois Mule, CableLabs; Vivek Badranith, France Telecom; Mike Jablon, Time Warner Cable and Jeff Pulver, Pulver.com.  During this member's only-event, the panel planned to discuss the current status of HD voice services to help the cable operator community assess their plans for HD in their networks.

Looking ahead, HD Voip will also be the topic of discussion at a number of other upcoming events, including:

ITExpo West in Los Angeles - "Pushing the Envelope with HD Voice - a State-of-the-Industry Panel Discussion" on Wednesday September 2nd at 10:30 AM

HD Communications Summit in New York City, an all-day event focused on HD in voice communications on Tuesday, September 15th in the New World Stages venue.

VON Fall 2009 in Miami Beach - "HD Voice: The Future of VoIP? A panel discussion on the impact of HD on our industry" on Wednesday, September 23rd at 12 noon.

I hope you will join me and participate in one or more of these sessions to contribute to this industry-changing discussion.

During this last "blogisode" we moved from a discussion about the technology and weakness of the SIP RFCs to what I feel is a far more difficult hurdle, the political challenges with SIP interoperability. As with national and global politics, getting people with diverse needs and existing investments to compromise is a significant challenge.

So how do we break the almost infinite cycle of continuous interoperability testing between the growing number of hardware vendors, software applications and carriers?

One approach that immediately comes to mind is to define a "gold standard" that each of the market participants can test against, and create a predictable level of interoperability assurance between systems. As one of my readers duly noted, this is the driving vision of the SIP Forum, and their SIPconnect effort, trying to make SIP Trunking interoperability more predictable. The SIPconnect Technical Recommendation 1.0 specifies the interface between a SIP-enabled Service Provider to a SIP-enabled Enterprise Network (aka "SIP Trunks"). The specification essentially narrows the hundreds of valid options within SIP to a defined set for one application, the replacement of legacy PRI circuits. So far so good, but don't get too comfortable quite yet.

The down-side of SIPconnect (or any other interoperability standard) is as a side effect, it limits the services and options available to users of the systems. Think SIP handcuffs--great as long as you aren't the one wearing them. Features like security, network call transfers, voice messaging, fax, hosted PBX functions, dealing with NAT and other areas were in many cases outside the 1.0 specification and thus not supported. So back to the drawing board, extending the SIPconnect specification to 1.1, which is an ongoing development effort trying to add these important capabilities without ruining the simplicity of the 1.0 specification.

But remember that SIPconnect is only focused on SIP Trunking and doesn't address interoperability between SIP systems within an enterprise or service provider. We've already seen cases where enterprises have one vendor for their SIP-based IVR and another vendor for their IP-PBX, and legacy TDM messaging systems, and are trying to integrate all these together. (Acquisitions and mergers have a funny way of making strange bed-fellows.) In this particular case, they couldn't even get the vendors of these systems to talk to each other, let alone actually fix protocol compatibility issues.

With pure SIP-based applications expanding in market share and perfectly good legacy TDM systems that are too new to replace, I believe intra-enterprise interoperability issues will become much more common. In the end, I envision that enterprises will need a on-premise network element from an independent vendor that will provide interoperability and security between their legacy systems, SIP systems and SIP Trunks they depend on for connectivity. Right now, it appears a Session Border Controller (security and SIP interoperability) and/or a Media Gateway (TDM interoperability) is the best way to solve many of the difficult interoperability issues facing SIP. Trying to get competing vendors to talk seems to be an exercise in futility.

Next we'll discuss who should be responsible for interoperability and security with SIP Trunking--the service provider or the enterprise?
 

 Earlier this week I shared with you a few thoughts on SIP Interoperability discussing what I felt where the root causes of incompatibility between two or more SIP-based systems.  I clearly hit a raw nerve with a few of you, flooding my email box with your own stories of interoperability issues.   You shared with me your own experiences with registration problems, call transfers, security, message waiting indications, even fax issues.  It seems the couple examples I gave were only the tip of the iceberg.

 

Let's move past the technical issues with SIP Interoperability and talk about a far more difficult challenge - the politics of SIP Interoperability.

 It appears to me that soon after the authors of RFC 3261 finished their work, the fun really started.  As the development teams of the various product and application companies started to build their solutions based on RFC 3261, the looseness of the specification allowed them to make wildly different choices all "within specification".   The result was that you had developers that had invested untold hours of hard work into developing a protocol stack that worked fine in their own lab and with their own products, but had serious interoperability issues with other vendors.  To each of the developers, it appeared that "everybody else screwed up". 

 So now you have a number of over-worked developers that would have to go back into their products and re-work significant parts of their SIP stacks - just because someone else made some bad choices.  The end result is a classic stand-off with each of the vendors saying "we followed the spec, you should change".  So much for "Open and Standard".

 

To make things even more politically complex, many of the vendors are starting to compete in the marketplace, vying for the same markets and customers.  In this competitive environment, interoperability is a double-edge sword.

 Okay, so let's pretend our developers get past their own stubbornness and decide to make some changes to be more interoperable.  Who do you do your interoperability testing with?  Do you test against anyone that comes along?  Or maybe just in cases where "the business case works"?   What happens if you or anyone else makes changes?  Do you re-test with everyone?  It was easy when there were just a few other applications to test with on the market, but now with hundreds of applications and devices to test, it becomes clear that the maintenance of SIP interoperability testing becomes a bigger burden than the original development. 

 So, how do we work around these political problems and break the cycle of continuous interoperability testing?  This will be the topic of my next post. 

 

For those of you that have deployed SIP-based solutions or SIP Trunking, there is a pretty good chance that you've had to navigate your way through the maze of SIP interoperability, wondering why it is so difficult to get a straight answer out of anyone on whether two systems will work together or not.

SIP is supposed to be a standard and eliminate many of the challenges with integrating systems from various vendors together, right? If my IP-PBX is RFC 3261 compliant and my SIP Trunking service provider is RFC 3261 compliant, they should just work, correct? Well--maybe or maybe not. Most likely there will be interoperability issues.

Today I thought it would be good to start a multi-part series of posts to explore why SIP interoperability seems to be as challenging as it appears, how the vendors are dealing with it and solutions you can use to solve difficult interoperability challenges.

The Root Cause
Let's start with a discussion on the root causes that make SIP interoperability difficult. The problem starts not with technology, but with the way that IETF Requests For Comments (RFCs) are developed. As opposed to the old ITU specifications that the Telecommunications Industry has lived with for the last four decades, IETF RFCs and Drafts are developed in an open and communal environment, using committees and consensus to craft the specification. This has very many positive benefits, but also a few predictable negative side effects. The problem is that RFC 3261 that defines SIP has become "everything to everyone" and bloated in both size and in flexibility.

Performing a simple word count on RFC 3261 yields some interesting insight into the problem:

Weak Terms
Can = 475
Option = 144
Should = 344
May = 381
 

Strong Terms
Shall = 4
Must = 631

As you can see, the number of weak terms "May," "Should," "Option" and "Can" outnumber the stronger "Shall" and "Must," which results in a very loose specification that allows the developers of SIP-based systems to make plenty of decisions on features of functions. The byproduct of this is that two systems can be completely RFC 3261 compliant and completely incompatible.

Examples
In our experience, there are no fewer than five "correct" ways to transport DTMF tones from one end point to another:

• In-band--leaves the DTMF tones in the RTP streams 
• RFC 2833--uses specialized payload packets in the RTP stream to indicate a DTMF tone 
• SIP NOTIFY--uses a SIP message to indicate the presence of a DTMF key press 
• SIP INFO (Nortel)--a technique frequently used in Nortel systems that uses a SIP INFO message 
• SIP INFO (Cisco)--a variation on the above, but with some slight modifications.

Another example that is causing a lot of grief right now whether to transport the SIP messages over UDP or TCP. The RFC indicates that either is okay and recommends supporting both, but few manufacturers actually do support both. The vast majority of equipment and application developers chose SIP-over-UDP. Microsoft chose SIP-over-TCP. Again, both are within specification and completely incompatible.

And this my friends is only the tip of the iceberg.

While these technical challenges may seem difficult to overcome, they can be solved. However, the political issues are another story. I'll discuss these in the next post.
 

• Reduction of trunk lines to branch offices
• Elimination of toll charges for inter-office calling
• Consolidation of trunking facilities
• Centralized applications (voice mail, IVR and ACD systems)
• Easier remote management and elimination of "truck rolls"
• Improved productivity with Unified Communications capabilities

• Survivability - what happens at the branch if the wide-area network goes down or accidentally cut? An extreme example of this is the outage in San Jose, CA on April 8^th, 2009 where a major fiber optic line was cut by vandals.
• Local Numbers - will you still have the same local numbers that long-established customers have on their prescription bottles or refrigerator magnets?
• E911 - if there is an emergency, will the first responders be directed to the right facility?
• Broadband Availability? - while quite common in urban and suburban areas, wireline broadband is either very difficult to get or very expensive in most of rural America. Common wireless technologies including EVDO, WiMax and satellite are not conducive to voice traffic and may insert significant latency or jitter.

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!

Over the last few weeks, I've spent quite a bit of time talking with a variety of partners about leveraging SIP in large enterprise deployments and specifically the architectures used to support branch offices.  When I first starting working on this problem, my original reaction was "Simple, just put in a softswitch and connect all the sites together via SIP - Done".  

It turns out it's not that easy.  From listening to our partners and their customers, I have learned there are a few real challenges they deal with when deploying communications systems into distributed branch office situations:

First you need to get good quality broadband to every one of your branch offices, which is hard to do once you leave the urban/suburban rings of most cities.  Getting voice-grade  broadband to remote offices in rural America can be very expensive and wipe out any potential cost savings.  Consumer grade broadband is easier to get, but even then not predictable enough for commercial applications.

Second is the question of reliability, which was recently demonstrated by the massive Internet outage in San Jose. What will happen to your business if the broadband connection to the site is cut?  Do you just close for the day and kiss off the revenue?

Third is network traffic optimization - does every call really need tie up your broadband service?  Is there a more efficient way to leverage the expensive and shared broadband that services the branch offices?

The solutions seems to be in an architecture that fits somewhere between the two extremes of fully centralized and fully distributed, but exactly where depends on the individual business.

To discuss these challenges and some potential solutions, I've invited Bruce Mazza, Director of Branch Office Solutions for Avaya to join me in a live webinar on Tuesday, April 14th at 2 PM.  I encourage you to Click here to register for the live event or listen to an on-demand recording of the event.

Yesterday's big news here at VoiceCon Orlando 2009 was the launch of Avaya Aura, a new SIP-based architecture and strategy from Avaya that  "simplifies complex communications networks, reduces infrastructure costs and quickly delivers voice, video, messaging, presence, Web applications and more to employees anywhere."

As Kevin Kennedy, president and CEO, Avaya noted in both his keynote address and press materials:  "We've seen some organizations use SIP routing to reduce trunking costs by 20 percent to 60 percent. With this new architecture, for the first time, the way we communicate is defined by the applications and the user, not the network."

Bruce Mazza, Director of Branch Office Solutions and Alon Waks, Solutions and Product Marketing for Avaya gave me a quick tour and demo of the Remote Branch portion of the solution showing how branch offices leverage  AudioCodes MediaPack analog gateways to provide survivability in situations where the wide-area-network were to fail or become unavailable.

In addition to survivability, the AudioCodes MediaPack gateways can provide connectivity to analog phones, fax machines and other legacy devices in remote branches.

This announcement highlights one of my key strategic visions of the power of SIP, connecting diverse and distributed businesses together at far lower costs than using the PSTN to call inter-branch.  

I'll be joined by Bruce Mazza to discuss the new Avaya Aura strategy and branch office survivability in an upcoming webinar that I'll be hosting.  Click here to register for the event. 

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?

Another interesting presenter and topic that I am looking forward to at the AudioCodes / ScanSource Solutions Theater at VoiceCon Orlando is Rick Q. Chin from Interactive Intelligence.  AudioCodes has been working with Interactive Intelligence since the very earliest days of their Customer Interaction Center, one of the premier SIP-based contact center solutions on the market.

AP: Rick, what's the title/topic of your presentation? 

RC: The presentation I'll be bringing is 'Improving the Customer Experience with CIC' and it discusses how to help companies win and keep customers

AP: How does Interactive Intelligence CIC help enterprises in this difficult economy? 

RC: Increases efficiency, effectiveness, and utilization of resources, improves customer service and increases customer satisfaction, provides lower operating and ownership costs

AP: Is there an opportunity for Value Added Resellers (VARs) that would be at the event to leverage with your solution? 

RC: Absolutely, Yes they would get a lot from my presentation.

 AP: What about the Interactive Intelligence strategy is different from before the economy went sour? 

RC: Focusing on communicating the value and high ROI of our completely line of applications, unique truly all-in-one product, and the higher productivity and effectiveness it provides vs. higher ownership and overhead and piecemeal solutions from the competitors.

AP: Who would get the most out of listening to your presentation? 

RC: A range of executives, Call Center Managers, Disaster Recovery Officers would get the most from the presentation, but we hope others will stop by to hear about improving customer care.

AP: Where would someone learn more about your solution?

RC: They can visit www.inin.com or give us a call at 800-267-1364

Rick's presentation is scheduled for Wednesday, April 1st at 2:30 EDT in the AudioCodes / ScanSource Solutions Theater in Booth #931.  Stop by and you could win an iPod Touch!
 

I'm very much looking forward to the upcoming Solutions Theater and iPod Give-away we are hosting at VoiceCon later this month.  As part of this series of presentations on SIP applications, we'll be joined by Bill Miller, Vice President of Product Management for Digium.  Bill brings a wealth of experience and knowledge on both IP-PBX solutions and open source to the stage.  Bill was kind enough to share with me some thoughts on his upcoming session: 

AP: Thanks again for presenting in the upcoming Solutions Theater at VoiceCon.  Can you give us a sneak peek at the topic of your presentation?

BM: My session is titled "Open Source Alternatives in a down economy"

AP: Please share with us how Digium|Asterisk and open source solutions help enterprises in this difficult economy?

BM: The business case for open source and open source based solutions is compelling for both business and technical reasons. We will explore why the momentum is not just building daily, but there are proven solutions, case studies and growing list of enterprise class solutions.

AP: How can Value Added Resellers (VARS) best leverage Digium|Asterisk?

BM: VARs are adopting alternatives to traditional telecom solutions at alarming rates. The ability to leverage experience already gained from traditional telephony solutions, the new emerging models provides new channels of revenues, new pipelines of opportunities and many alternatives to build custom telephony solutions to grow faster than the market. price conscious users want to kick the tires and understand open source today.

AP: Who would get the most out of listening to your presentation?

BM: Resellers and enterprises or all sizes

AP: Bill, where would someone learn more about your Digium Open Source solutions?

BM: The Digium booth is #1328 here at VoiceCon and our web site is www.digium.com. To learn more about the open source asterisk project, visit www.asterisk.org.

Bill's presentation is scheduled for Tuesday, March 31st at 4:00 PM EDT in booth #931 - see you there and good luck winning the iPod Touch!

 As noted in yesterday's post about VoiceCon, our opening speaker in the AudioCodes / ScanSource Solutions Theater will be Joel Maloff, Vice President of Marketing for BandTel.  BandTel is a pure SIP Trunking service provider that utilizes AudioCodes gateways to connect their services to end-customer TDM PBXs or other equipment.  Joel was kind enough to spend a few minutes with me yesterday, providing a preview of what to expect in his presentation at VoiceCon.

AP: Can you share with us the title and topic of your presentation?

JM: Sure, the title is 'SIP Trunking - Ready for Prime Time '

AP: How does SIP Trunking help enterprises in this difficult economy?

JM: SIP trunking has two immediate benefits for enterprises, especially in light of economic conditions. The first is the ability to broaden an enterprise's reach via the use of remote local telephone numbers. For example, an enterprise that is forced to consolidate and close down physical locations can retain those telephone numbers and have them ring into one or more centralized facilities. In this way, local identification is retained, operations are consolidated, and costs can be dramatically reduced. In addition, new locations can be opened throughout North America and internationally with those calls received by an enterprise call center. The second benefit is the ability to reduce reliance on dedicated telephone circuits for voice and reduce the cost of incoming and outgoing telephone calls. This includes both domestic and international. Our research indicates that savings of 25% to 70% of the annual telecommunications budget is achievable. In difficult economic times, those savings go directly to the bottom line and can be reused to the benefit of the business.

AP: How will value added resellers (VARs) benefit from reselling SIP Trunking from BandTel?

JM: BandTel offers a percentage of monthly recurring revenues brought to us for the life of the service. As long as the customer remains with BandTel, our partners will receive the benefit of their efforts.

AP: What about the BandTel strategy is different from before the economy went sour?

JM: Our strategy has not changed. We believe that businesses want to do more for their operating expenditures, and to find new ways of serving their markets. What has changed is the mindset of our customers. They are no longer content to pay high prices to traditional communications carriers without considering new and different ways to accomplish the same results for less money.

AP: Who would get the most out of attending your presentation?

JM: Anyone who is responsible for finding ways to lower communications costs without sacrificing value will find the presentation valuable. This includes information technology and telecommunications staff organizations, financial personnel responsible for scrutinizing budgets, and value-added resellers seeking an additional profit center to the benefit of their customers and clients.


AP: Where would someone learn more about your solution?

JM: BandTel's website - www.bandtel.com - offers a wealth of information including white papers, FAQs, and case studies.

Joel's presentation is scheduled in the AudioCodes / ScanSource Solutions Theater in Booth #931 at 4:00 PM EDT on Monday, March 30th.  

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!