Pardon my exuberance but I am about to write about a product that I have been beta testing for the past few months. Seven months ago I knew very little about small business PBX’s other than to know they had big $$$ signs associated with their operation.

Back in 1992 my employer of the day paid $28,000 for a PBX that would service our local Canadian office of six to ten employees – and all the extensions had to be physically in the same office. Of course we would also be paying significant long distance charges, especially for out-of-country calls. The same employer was paying about $10,000 per month for a T1 line between California and their European office in Ireland. The economics of worldwide communications has changed somewhat since then; today’s announcement from In Store Solutions of the FREETALK Connect small business PBX takes small businesses further down that path.

About six weeks ago, while visiting In Store Solutions head office in Spain, using Skype for iPhone in a WiFi zone at my hotel room, I called a SkypeID that gave me the options to press 1 for Jim, 2 for another person and 3 for another person. I pressed 1 and a few seconds later my BlackBerry was ringing me back in Spain, But the PBX was at my home near Toronto, Canada as was an attached Linksys VoIP phone that the PBX also rang simultaneously with my BlackBerry in Spain. Had I answered my BlackBerry the only charge would have been my voice roaming package $1.25 per minute. Had somebody else originated that call in the way I did, I also would have had the option to answer the call, with no resulting charges for either party, on my PC’s Skype client which at the time was also in my hotel room in Spain¹.

Bottom line: any Skype user worldwide could have reached me via this PBX on a “local” or “remote” phone, a mobile smartphone or my Skype client with minimal, if any, costs for the call. And, in practice, geography as a factor to Find Me or Follow Me was not in the equation.

Today In Store Solutions announced FREETALK Connect, a small business PBX that changes the algorithm and business model for small business voice communications. With FREETALK Connect a business can benefit from:

• accepting inbound calls made via Skype, a Skype Online number, a SIP client, Asterisk (IAX2) or the PSTN, via either an analog line or a VoIP service provider.
  • incorporating Find Me, Follow Me features
• accept calls on local analog phone lines, local or remote VoIP phones, mobile phones anywhere or remote PSTN-accessed phones – all provisioned within the PBX.
• placing outbound calls over Skype, SkypeOut, a SIP-connected VoIP service or an analog PSTN line.
• managed voice mail including consolidation of voice mail and email into one mailbox

As I experienced the different communications paths this product opened up, I realized that this box can revolutionize how small business communicates with their suppliers, customers and business partners – worldwide.

The press release lists all the services that will be on board the FREETALK Connect when it becomes available for purchase by U.S. registered Skype users in March on the Skype Store:

Some of the key features from the Jazinga platform found in the FREETALK Connect include:

• Callback / Dial-around
• Access to Skype Buddy lists
• Auto Attendant / IVR
• Paging
• Call Parking
• Remote Extensions
• Music on Hold
• Conferencing

The FREETALK Connect also has an easily configured and updated:

• Managing routes to users, telephone services, and applications
• Providing SIP/Skype telephone service management
• Router management (networking, port forwarding, DNS, DHCP)

“Jazinga’s products consistently ensure call integrity by integrating quality of service and prioritizing voice traffic on the network into an affordable, simple product,” said In Store Solutions COO Craig Smith. “There was no question that FREETALK wanted to partner with Jazinga to develop the FREETALK Connect, because it continues our goal of working with the best providers to distribute outstanding products around the world.”

“FREETALK Connect is designed for small businesses with between 2 and 49 users, an undersold market that desperately needs UC functionality,” said Randy Busch, CEO of Jazinga Inc. “As a result of our partnership with In Store Solutions, the telecom technology playing field is much more level between larger enterprises and their smaller competitors.”

In terms of the technology within FREETALK Connect it incorporates both Skype for SIP and Skype for Asterisk to provide an optimum level of Skype-based services but at the same time providing inbound calling flexibility and optimizing outbound calling expenses, all combined with the call integrity features quoted above.

FREETALK Connect will requires users to have a Skype for Business account, available through the Skype Business Control Panel, associated with each unit. More on this aspect of FREETALK Connect is the subject of a future post.

Bottom line: Targeted at small businesses with 2 to 25 employees and/or business partners with its wide range of inbound and outbound calling options, an attractive price point, incorporation of Skype for Business infrastructure and the ability to self-configure via wizards, FREETALK Connect represents a game changer in the small business communications market. Lots more will be written about this product as the applications experiences evolve.

¹In this situation I could not have the Skype client answering option since I was using my Skype account to originate the call.

Full disclosure: In Store Solutions has become a client of Denali InterConneXions, publisher of Voice On The Web, building on the author’s previous business development experience with establishing partnerships that can assist with the promotion of a primary vendor’s offerings. A more complete statement will follow shortly.

Other posts:

Saunderslog: FREETALK Connect, the worlds first Skype-enabled Unified Communications system.

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Building on Skype’s recent announcements of its evolving Skype for Business activities, Chief Strategy Office Christopher Dean gave a keynote address yesterday at a joint VON/Channel Partners Conference session. Three posts have followed up on the session:

Khali Henderson, in Skype Not Just Telecom’s “Bad Boy”, CSO Says, reports Christopher’s overview of their approach to business markets:

“Skype is no longer just sort of the ‘bad boy,’ disruptive influence in the telecommunications market,” he said. “What we offer is strong opportunities to leverage these trends for operators and their customers.”

The company, he said, is actively seeking partnerships with service providers – wireless and wireline operators and MSOs – as well as OEMs and ISVs and applications developers, to be part of what he calls an “emerging ecosystem.” Skype’s role is to be an overlay of services – audio conferencing, video, desktop sharing, presence, IM, file sharing and more – that drive demand.

Khali reports that Christopher went on to discuss the success of Skype’s partnership with the Hutchison 3 Group and the services offered involving 3’s Skypephone as an example of how Skype has helped a partner successfully build their business.

Alec Saunders reports on the session in Skype counters critics, embraces ecosystem:

…. Skype CSO Chris Dean began his remarks at VON by saying that there is an ongoing sea change in the telecom market, and it’s only accelerating.  The macro trends are around wireless, wireline, emerging applications and service models, the economy and the regulatory environment.  And he warned that there are immense risks to operators that don’t adapt.

Jon Arnold, in VON – Final Day/Skype Keynote – Seeking Channel Partners, reports:

Most of us know this [the business market] is a top priority for Skype, and the most newsworthy item from the keynote was their new focus on channels, which of course ties in nicely with both events here – VON and Channel Partners. The main idea here is that Skype will be launching a service provider VAR program later this year. Christopher provided a URL about the program for reference, but as you’ll see, there’s not much there yet.

So, in the course of a few minutes, the audience heard about how Skype is looking to cross the chasm and partner with service providers. That creates all kinds of interesting scenarios, and Christopher pointed out how this is part of their bigger vision to be more open and partner-friendly in the post-eBay world. On that note, he didn’t have much to say about the Extras program other than they understand how important it is, and are overhauling it now. So, it’s not quite dead yet….

Getting out a message differentiating the Extras program as a partner marketing program, which is coming to an end, from the evolution of any forthcoming Skype developer program has been difficult challenge for Skype. I suspect Christopher was really referring to Skype understanding how important a well-articulated and effective Skype developer partner program is. Reference to the Skype Service Partner Program was made in yesterday’s post: Skype for SIP: Getting Legitimization, Endorsement and Traction.

Bottom line: Christopher’s keynote is one more sign that Skype is getting serious about its approach to the business market, building on the recent launch of Skype for Asterisk and Skype for SIP. A significant challenge will be, once Skype is embedded within business communications infrastructure as a voice calling option, to evolve such that this infrastructure can then take advantage of its presence, chat and video infrastructure.

But perhaps more important: For over six years consumers and many businesses have experienced Skype as a lowl cost or free voice, chat and, more recently, video conversation service – with minimal support from Skype itself. Skype is finally gaining acceptance, certainly now at the communications vendor level, as a viable and reliable alternative to legacy PSTN services. And Skype finally appears to be building the resources and programs required to properly support business communications operations.

This trend recalls the early days, 25 years ago, of the PC itself when adoption of the PC as an enterprise resource only came about because employees were bringing PC’s into their offices on an individual basis and demonstrating the resulting productivity and cost effectiveness to their management. Been there; done that!

Skype VON 2009 Keynote Presentation – Christopher S Dean

View more documents from SkypeConversations.

Thanks to Jon Arnold for providing the pictures, with permission.

Saunderslog.com:  Virgo’s first VON a solid outing

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This week about 50,000 people have gathered in the northern Nevada desert for the annual Burning Man ritual. You really have to visit their website to get a feel for what it is about and the principles that guide this annual one week experience.

Started in the mid-1980s, Burning Man is designed to generate a society that connects individuals to their creative powers, participation in community, civic life, and the world of nature. Gathering on an ancient lake bed, about 140 miles north of Reno, participants decide how and what they’ll contribute to the community.

But one guarantee – neither AT&T nor Verizon, neither Sprint nor T-Mobile had a cell tower within the horizon of the Black Rock City playa. As a result, working with Voxbone’s iNum service, up to 10,000 participants can receive SMS messages or, within capacity limitations, receive voice calls at country code +883, iNum’s worldwide country code accessible for free from many partner service providers and via local access numbers in 45 countries worldwide.

So, with no coverage by the commercial wireless carriers, how does this connect? Voxbone has worked with the Open BTS Project:

…. an effort to construct an open-source Unix application that uses the Universal Software Radio Peripheral (USRP) to present a GSM air interface (“Um”) to standard GSM handset and uses the Asterisk software PBX to connect calls. The combination of the ubiquitous GSM air interface with VoIP backhaul could form the basis of a new type of cellular network that could be deployed and operated at substantially lower cost than existing technologies in greenfields in the developing world.

In plain language, we are working on a new kind of cellular network that can be installed and operated at about 1/10 the cost of current technologies, but that will still be compatible with most of the handsets that are already in the market.

The OpenBTS team has constructed a cell tower at Black Rock City, that, in the “leave no trace” principle of Burning Man, will be taken down at the end of the event on Labor Day (next Monday). Effectively working together, Voxbone and the OpenBTS project have demonstrated how to set up and install at a low cost a complete GSM-based wireless service in remote and otherwise unserviced regions of the world. Full details are here but it is interesting to note:

• Who is doing this? This is an activity of the OpenBTS project. The entity legally responsible for the test is Kestrel Signal Processing, Inc.. The actual on-site engineers are David Burgess and Harvind Samra.
• Is this legal? Yes. We have an experimental license, WD9XSP, and a consent letter from Verizon authorizing us to operate in their KNKN224 block A cellular spectrum. We are also coordinating operations with Commnet Wireless to prevent interference.
• How much does the service cost me? Nothing.
• What do you do for emergency calls? Nothing. We are an experimental service and do not support emergency calls.

and

• My phone shows service but I can’t make a call. Why not? Only the first few hundred subscribers will get full speech service. Everyone else will just get SMS. There are two good reasons for this:
  1. We don’t have enough equipment for everyone to get normal speech service.
  2. Widespread speech service would change the nature of BRC in ways a lot of people would not like.

• Will my carrier charge me for every text message I send or receive on the Papa Legba network? No, the Papa Legba network is a separate isolated network from your normal carrier’s network, and no charges will apply as no SMS is being sent or received through your normal carrier.

Note that “Voxbone is providing funding and most of our VoIP connectivity”.

OpenBTS Chronicles: We Loaded the Truck Today and co-founder David Burgess reports on getting it going:

So we turned on the PA for the first time today and immediately got swamped with registration requests, which is actually normal and a good sign given our operating conditions. We also got a report of someone receiving a text message at Center Camp. That’s only about 1 km, but the impressive part is that the guy with the phone was in a
sealed shipping container (the BRC NOC) at the time. We were delayed seriously by high winds that made tower work dangerous, so a 4-hr tower job took just over 11. But it’s up thanks to the heroic efforts of the rigger. It was a little scary to watch.

From the Voxbone press release:

Voxbone worked with the OpenBTS Project to automatically assign iNums to cell phone users at Burning Man, making them instantaneously reachable from networks such as Skype and Google Talk.

“We are showcasing how iNum can be used to connect remote or unconnected areas of the globe with the rest of the connected world, using cell phone infrastructures such as OpenBTS,” said Voxbone CEO Rod Ullens.

“In addition, we are demonstrating to people attending Burning Man that they can receive voice calls and text messages from services like Google Talk on their cell phones at the festival. Burning Man is a great event at which to highlight these types of services because the attendees are open to new ideas and new experiences, and we are enthused to be a part of that.”

The iNum service is provided free to users at Burning Man. At the same time, it will enable users to spread the word about how valuable iNum is when it comes to connecting the unconnected.

“In partnerships like this one for Burning Man, there is an opportunity to demonstrate a technology for quick and cost-effective connectivity to regions that urgently need it,” said OpenBTS Project co-founder David Burgess. “In this case, the connectivity is a convenience to Burning Man participants, but iNum also can play a critical role when war, natural disasters and other emergent situations require quick communication.”

Alec Saunders, in OpenSource Meets GSM at the Burning Man, asks “Why do this?”

Naturally it’s a nice showcase for VoxBone’s iNum technology.  More to the point, OpenBTS believes that by making inexpensive mobile base stations available built around open source solutions, telephone service can be made affordable even in the remotest parts of the world.

Would be interesting to hear how it turns out in terms of number of cell phones connected and conversations handled. but the project is certainly a contributor to the overall spirit of innovation and creativity encouraged by the Burning Man ritual.

Question: while Skype for Asterisk was just released this week, can we expect the OpenBTS Project to add a Skype for Asterisk channel to their PBX next year?

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Following a link from Dan York’s update to his post outlining technical details of Skype for SIP, I found a link to this post from IfByPhone’s Irv Shapiro: Why Skype for Asterisk is more important than Skype for SIP.

First, Irv has done perhaps the best articulation yet differentiating Skype for Asterisk from Skype for SIP:

Skype for Asterisk, which is still in closed beta, is a true Asterisk channel driver. This allows Asterisk based solutions to make, receive and transfer Skype calls. A significant capability of the SFA solution is its support for terminating a call to a Skype user name, for example a PC based user of the Skype client.

Skype for SIP is a very different animal. This service provides VOIP trunk support for existing SIP based PBX systems, which may include Asterisk. Unlike SFA where calls may be placed to any Skype user, SFS calls may only be terminated to PSTN end points.

Irv goes on to provide his view on how this plays in the Voice/Telco 2.0 marketplace. Then he provides an example of where he sees the benefit of SFA for hosted services:

From our perspective as a cloud telephony company, providing hosted telephone applications, SFA is much more interesting. While either service would allow us to provide IVR services to Skype users, the SFA Asterisk channel driver architecture allows us to terminate calls into call centers with no PSTN transport. Each call center Agent would just utilize a headset connected to a computer running the traditional Skype application. Customers calls would be able to originate from either a PSTN device or a Skype client, then route through the Ifbyphone IVR infrastructure and terminate to a call center via Skype’s computer to computer transport. This has the potential to change the cost structure associated with supporting call centers. A Skype based call center would not require a PBX or for even any centralized telephony components. The call center agents could be virtually located anywhere in the world on high quality Internet connections.

Keep in mind that calls terminated on Skype, as demonstrated on the 3 Skypehone service, have no associated termination charges. Long time Skype-based call center provider OnState has already demonstrated this business model; their hosted OnState Call Center service eliminates the need for a call center PBX. While a (prospective) customer can call in via several modes (Skype Online number, DID numbers, 800 numbers, Skype or click-to-call from a website), all the calls are terminated on Skype via their agent client.

Andy Abramson at VoIP Watch has, in a reverse manner, demonstrated this cost advantage also in his post, All Google Voice Needs Is a Little More SIP and Skype’s Game Changes, where he points out that, if Skype for SIP did terminate calls on a Skype client (i.e. – supported SIP—>Skype), then one could terminate Google Voice calls at a Skype client. But, as I have commented on that post and included in Skype for SIP: Sorting Through the Issues, Skype for SIP supports three types of connections and “SIP—>Skype” is not one of them.

Irv demonstrates that Skype for Asterisk has some interesting potential; it’s simply going to require more creative innovation that goes beyond simply making call connections through a SIP interface. It’s a matter of invoking Alec Saunders Voice 2.0 Manifesto which points out that the value lies in the applications.

The Voice On The Web public chat has continued to be lively through the week. Not only has Skype’s Senior Product Manager responsible for Skype for SIP, Chris Moore, participated but a couple of other guests have joined the discussion to clarify some issues.

First to summarize Chris’ comments:

• He is sorting through the overwhelming number of beta program applications with the intent to make accepted accounts live early next week
• G.711 codec support is in the late stages of development and may be available as early as next week; this will be accompanied by a list of countries where G.711 is supported.
• Chris has been taking input from participants with respect to SIP header information that would make Skype for SIP more broadly applicable. This, of course, leads into a discussion of going outside the “box” of SIP header standards. (SIP headers basically provide information that is transferred across a SIP connection.) More information from Jason Goecke at Adhearsion; Dan York at Voxeo has made an initial suggestion over at Github.

However, more significant have the the responses from Digium about their perspective on where Skype for Asterisk is positioned and the participation of David Tang, Vice President, Global Marketing at VoSKY in the Public Chat discussion.

Steve Sokol at Digium put up a post, “The Rumors Of Our Death” where he outlines key differentiators of Skype for Asterisk from Skype for SIP (slightly revised by Steve after feedback from Skype and chat participants):

• SFA can handle incoming Skype calls directly from any user on the Skype network.  SFS can receive incoming calls from Skype users names that are statically mapped to a Skype name to a SIP account.
• SFA can place calls to any user on the Skype network.  SFS cannot place calls to Skype users.
• SFA includes support for Skype presence information.  SFS has no support for presence.
• SFA includes buddy list management.  SFS has no buddy list management features.
• SFA calls are encrypted from end-to-end while SFS calls are delivered to the SFS endpoint devices (PBX) as unencrypted RTP streams.
• SFA supports multiple media codecs including G.711 aLaw and uLaw as well as G.729.  Wide-band audio will be available in a near-term revision.  SFS supports only compressed telephony-grade G.729 media streams.

Ultimately, the key differentiator is that SFA is designed to allow developers to build rich applications that are deeply integrated with the Skype network, while the SFS offering is a trunking solution for legacy IP-PBX systems that support SIP. Both have their place and neither is specifically competitive with the other. Both Digium and Skype are dedicated to Skype For Asterisk and both are working hard to make sure that when released it exceeds expectations.

Steve then goes on to discuss the current status of Skype for Asterisk, why it is taking so long to come out of beta and makes this important point about changes in the Skype Business Control Panel:

Enter the Skype Business Control Panel (BCP).  Skype’s Skype For Business division has launched an all-out effort to make business customers a corporate priority. Currently the BCP is simply a way to aggregate and share Skype credit.  As business services like SFA and SFS are rolled out it will take on a much more prominent role as the administrative interface for commercial customers.

The next update to the BCP will allow business customers to create master accounts that are owned and managed by companies rather than individuals.  With the BCP, Bob’s account would have belonged to AGI rather than to Bob, much like his AGI email account and company credit card.  If Bob went in search of greener pastures the system administrator could simply redirect his calls to another sales guy.

Jason Goecke has then put up his second post on Skype for SIP, Skype for Asterisk DOA; long live Skype for Asterisk, clarifying his positioning of the two offerings and concludes with:

The point is not to declare SFA dead, but to highlight these points in the hopes that the opportunities for deeper integration and the associated benefits are clear. For SFA to overshadow the SFS announcement, I contend these features are needed:

• Two-way chat via the Manager API
• Extended presence support
• SILK-codec support, which would require end-points (ie – SIP soft phones/hardware phones) embedding SILK

I understand that Digium and Skype need to get SFA to market, but a clear roadmap of what comes next and the timing would be immensely useful as developers look to place their bets. I applaud the work that Skype and Digium have been doing and the opening of the Skype network on multiple fronts. I just hope this is an opportunity to open the Skype network that little bit more.

David Tang at VoSKY has also contributed to the discussion; his main point:

We have our fair share of scars from being an early pioneer working with Skype.  Digium is just beginning to experience it.

From the tone of a few comments re VoSKY in the chat and considering their long experience with Skype implementations in businesses, a large reseller network and their creative ingenuity, we can expect that VoSKY will figure out a way to  go beyond simply continuing to be a “Skype trunking” solution for legacy PBX’s lacking SIP support (non-SIP PBX’s).

Back to the Business Control Panel (“BCP”) and its role. To date it has simply been a means for an administrator to manage Skype accounts in terms of handling Skype credits and implementing Skype on PC’s within an enterprise. Going forward Chris Moore has made the following comment:

I think I’ve chatted about the BCP in some interviews, yes the BCP will take a larger role in administration for small business of existing and new services. It will be the central portal for an SMB to use Skype.

Seems that there are many of the Voice On The Web Public Chat who are ready to start experimenting and developing with Skype for SIP; I look forward to monitoring the feedback as they finally get their hands-on experiences. Bottom line: from these developers, Digium, VoSKY and others, Skype has launched a significant opportunity to see innovation at work solving real business communications issues.

Overall Skype is starting to execute on their Business strategy through the new Business marketing group. The Business Control Panel is central to this strategy. We now have seen three activities in that direction: Skype for SIP, Skype for Asterisk and the Lotus Live support.

Update: Chris Moore has confirmed that the first beta account acceptances have gone out today; login details and passwords will be issued Monday/Tuesday next week.

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Yesterday’s announcement of the Skype for SIP beta created lots of commentary in the blogosphere; it remained the leading topic on Techmeme for several hours.

• Phil Wolff over at Skype Journal, in “Skype For SIP: Big Money, Skypeless, Brand Destroyer”, outlines some of the positioning with respect to Skype for Asterisk, channel schism issues and worries that it abandons Skype’s central tenants and could undermine the brand.
• Alec Saunders, in “Skype embraces SIP; announces beta service for business users”, talks about the impact on businesses using legacy PBX’s, the increased pressure on international long distance markets and a reminder that about 1/3 of all current Skype traffic is business related.
• Jason Goecke at Ahearsion, a Skype for Asterisk beta developer, worries about the impact on the market for Skype for Asterisk in “Skype for SIP = Skype for Asterisk DOA?”

But most interesting was the Voice On The Web Public Chat session that ensued. Being on a vacation I was away from my laptop for about seven hours (yes, it was a great day to be golfing) but when I returned I found a session that not only provided the flavor of the vibrant debate going on but also many answers. The key – Chris Moore, Skype’s Senior Product Manager responsible for Skype for SIP joined into the discussion to make several points:

• Most importantly – this is Skype for SIP beta; his team is looking for feedback from a range of beta testers, not only third party developers but also the IP PBX vendors themselves. To quote Chris:
  

  Yep, whole idea is to get feedback, see what people want/don’t want to help prioritize development of new features. Our group, Skype For Business, is getting a lot of focus and its a key strategic focal point for Skype now. We’ll never get it completely right at the start, but a start is still a start

• To qualify as a beta tester you need to have some experience with configuring IP PBX’s that are currently in the market. For instance, “you must have, and be able to configure, your own SIP Enabled PBX that supports G.729 codec.”
• Beta testers will need to create a temporary Skype name to use in the beta program; the issue of a separate SkypeID and how it was handled was hotly debated.
  One point to clarify:

  Skype for SIP actually can allow multiple Skype IDs to be mapped through to a single SIP identity. However, during the beta period, we are only supporting a 1 to 1 mapping due to the fact that Skype’s advanced Business Control Panel (BCP) configuration portal is not yet available at this time. If a company wishes to configure multiple Skype ID’s mapped to a SIP identity, we can accommodate this for a subset of the beta trial participants.

• The key connections made by Skype for SIP are:
  

  - PBX –> SkypeOut (PSTN)
  - SkypeIn/Online numbers (PSTN) –> PBX
  - Skype Clients –> PBX

• The business model, including pricing, has yet to be worked out. It will require Skype to determine a means to separate out business users from “consumer” users on both Skype for SIP and Skype for Asterisk. The principle here is that business benefits from the various features of these offerings; where such significant value-add exists there is a market for the services. But any pricing for either offering is yet to be determined, keeping in mind that both offerings’ desire to make significant market penetration will be a key moderator on pricing. Again quoting Chris:
  

  no new road is easy and enhancing our focus so Skype can support business users as a distinct group differently from consumers takes some work

• Differentiation from Skype for Asterisk will be an ongoing challenge. Initially Skype for Asterisk, which will be an Asterisk channel, includes handling SIP—>Skype calls, Skype-to-Skype calls and will incorporate presence information. (My comments on an ignored aspect differentiating these are below.)
• While SFS may incorporate some “Skype trunking” capability, SFS also offers the handling of inbound calling from Skype Online numbers.
• As for the current status of SFS, Chris says the focus at the moment is on getting voice connections right as this is the feature most demanded in the market:
  

  and this is baby steps for Skype, there are many more UC services to come, voice for SIP is the biggest ask from customers. Video/presence etc only make  sense when codecs are common between skype clients and SIP endpoints.Some work left there!

• Codecs are an issue. The initially supported G.729 codec requires licensing. The Skype for Business team is working on incorporation of G.711 within a few weeks but eventually hopes to assist with the adoption of the SILK codec across the industry by supporting SILK. But this also requires SILK support at the end points. Bottom line: codecs have both a technology and business issues to be resolved.
• Security is an issue that will probably provide one of the major challenges as Skype opens up access to its cloud from both Skype for Business offerings.

My own observations, not having been around for the discussion:

• Skype for SIP and Skype for Asterisk: the discussion yesterday centered on technical issues but totally ignored the market channel issues.
• Skype for SIP requires working with the various IP PBX vendors who have their own distribution channels; Skype for Asterisk is a partnering project with Digium who has its own unique distribution channels. These channels are both looking for innovation that keeps them in business. Yes, the common issue is that SFS and SFA both threaten the long distance market, but the VAR’s and System Integrators in these channels will need new offerings that meet this challenge.
• A significant challenge for both SFA and SFS will be the User Interface. How much will either of these products make the installation,  configuration and operational aspects so user friendly that outside support requirements are minimized or eliminated. This is a huge challenge in the SMB market as many small business owners will simply not want to be involved with such issues; they have their own business specialty to focus on. The value-add of VAR’s and SI’s serving any business market comes in making business communications low cost, seamless to the business operation and five 9’s uptime for their customers.
• There is not one solution that will meet the complete world of business needs. Currently in this SMB space we have several vendors: Cisco, Microsoft Response Point, Avaya, Siemens, etc. The world would be up in arms if there were only one offering for providing the disruptive innovation for which Skype can provide support to these channels.
• The issue of addressing consumer users vs business issues recalls the PC market twenty years ago when PC hardware vendors, initially Compaq, finally figured out how to differentiate and address these markets. This was a major issue I had to deal with in setting up a PC reseller network at the time. Today we see Dell addressing these various markets as soon as you launch into their website.

The Skype-enabled world is only healthy if there are alternative offerings and channels. Recall the theme of Skype President Josh Silverman’s reminder to the analysts last week: “Skype is a software company but we plan to be everywhere”. The Skype for Business group is focusing on making the software required to support these offerings; this software will leverage the IP PBX vendors and Digium distribution channels to drive adoption through into the various regional business market channels worldwide.

Update: While Om’s post, Skype Now Means Business, Friends the SIP World, demonstrates that he has not been following very closely the strategy being laid out by Skype’s new executive team over the past few months, the comment stream it triggered  is well worth reading. For instance, check out SFA developer Tim Panton’s comment about the opportunity for innovation he has found in SFA:

That aside, I’m a huge fan of S4A and our experience to date has been positive. We leverage the improved identity management (vs PSTN or SIP) and low costs of Skype to do some very cool integration with business processes and web applications by routing Skype calls through the extensible and scriptable platform of Asterisk. [author’s bold]

Next Post: Why Skype needs to get Skype Public Chat back into the latest release of Skype for Windows 4.0. The discussion referenced above demonstrates that this is the major failing of this version.

Update: Andy Abramson at VoIP Watch: Skype and SIP Equals A Takeaway of Toll Free

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• Skype Taps ShoreTel for Enterprise Play (gigaom.com)

Skype announces launch of a beta version of Skype for SIP, connecting IP-PBX’s to the Skype cloud.

Dan York, while a keen supporter of Skype, has been campaigning for a Skype to SIP interconnect on his Disruptive Telephony blog. In a post last fall, “Skype and SIP – the two sides of the issue raised by Michael Robertson”, he says:

We need to build the interconnect.

Yeah, there are a TON of issues out there that we still need to address to build that interconnect. There’s a whole host of security issues… there are billing issues… there are trust issues… there are network plumbing issues. Yes, there are all those issues. But if we are to succeed in ultimately bringing about the rich communication experience we want, we need to make this happen.

And for that, Skype’s walls need to come down…. at least a bit.

What we need is that Interconnect from Skype’s cloud out to the emerging IP infrastructure. Think about it… Skype right now has a two-way interconnect between Skype’s cloud and the cloud we know as the PSTN. It’s called “SkypeOut” and “SkypeIn” (or whatever marketing names they are being called now). If you dial my SkypeIn number, you can reach me on Skype wherever I am. From my Skype client, I can call anyone on the PSTN. The two-way interconnect is already there.

This morning Skype announced the launch of a beta version of Skype for SIP as a major initiative for further penetration into the business market. Basically Skype for SIP allows SIP PBX owners to access the Skype cloud as additional inbound lines , via Skype Online numbers, and outbound extensions to their existing PBX, via SkypeOut. From the press release:

Key Features

The beta version of Skype For SIP will enable business users to:

• Receive and manage inbound calls from Skype users worldwide on SIP-enabled PBX systems; connecting the company Web site to the PBX system via click-to-call
• Place calls with Skype to landlines and mobile phones worldwide from any connected SIP-enabled PBX; reducing costs with Skype’s low-cost global rates
• Purchase Skype’s online numbers, to receive calls to the corporate PBX from landlines or mobile phones
• Manage Skype calls using their existing hardware and system applications such as call routing, conferencing, phone menus and voicemail; no additional downloads or training are required.

In an interview with Skype Senior Product Manager Chris Moore, I learned:

• Skype’s’ goal will be to have a totally user friendly interface such that Skype for SIP is a seamless service
• Administration will be done through an extended Skype Business Control Panel interface which, amongst other activities will establish Skype for SIP account mapping, manage online numbers and perform appropriate configuration for the various PBX’s.
• Skype will be looking to work with the various SIP PBX and Unified Communications vendors, such as Cisco, Avaya, Nortel and Siemens. When I asked about Microsoft Response Point, which had been demonstrated to me the previous day, Chris confirmed, yes, they were another prospective vendor to be approached.
• during the beta, Skype will be used on a pay-as-you-go basis; pricing for the service once it is launched has yet to be determined
• Skype will be including security issues as one major consideration (I expect Dan York to have more to say on this issue)
• The service will be configured to take advantage of Skype’s royalty free SILK codec; if both endpoints of a call support SILK, then the SILK support will carry across the Skype for SIP interface.

This has to be a landmark announcement for the IP-based communications space. It effectively allows businesses to integrate the Skype cloud into existing infrastructure to handle all long distance and international calling. Some other initial observations:

• Does Skype for SIP impact the space currently served by Vosky’s PBX services?  It definitely leaves open for Vosky the analog PBX market; one question will be how does Vosky figure out a way to work Skype for SIP into their business activities and offerings during the six- to twelve-month beta?
• It’s currently a beta; applicants for beta participation can apply at www.skypeforsip.com. From the press release: “Applicants will need to be businesses, have an installed SIP based IP-PBX system, as well as a level of technical competency to configure their own SIP-enabled PBX. The initial beta is available to a limited number of participants.”
• Skype also expects to determine an appropriate distribution channel. While ease-of-use is a key goal, there will still be small business owners who will want third party expertise to manage their communications activities. So expect Skype to look to a VAR or Systems Integrator channel to support implementation.
• Skype for SIP broadens Skype’s penetration into the small-to-medium business market. Where Skype for Asterisk introduces a Skype feature-rich Asterisk channel to Asterisk PBX’s, Skype for SIP simply brings Skype cloud connectivity to a major subset of existing small business PBX installations.
• Along another feature axis we see hosted services vs on-premise implementations. Will widely geographically distributed small businesses favor a hosted service, such as OnState’s Virtual PBX, or one of these on-premise PBX offerings.

One sure call: the market for business long distance, especially international calling, is about to be radically disrupted as these services come into play. It brings about new business opportunities and, most significantly, makes any business a global business, if the will is there. With probably over one million IP PBX’s out there (Skype’s press release says over 438,000 were sold in 2008 along), Skype for SIP brings the slow but sure death of legacy international calling, one PBX at a time.

And, of course, Dan York brings his deep perspective, not only on how Skype for SIP works, but also on the security and codec issues surronding Skype for SIP: Skype tears down more walls with “Skype For SIP”.

Over the past three days there has been a continuous stream of presentations that provide insight into a wide range of issues, opportunities and perspectives in the IP-based communications space going forward. While there will be more posts covering individual presentations, it’s worth taking time to mention a few highlights:

• Most innovative, outside-the-box, presentation: Ge Wang of SMULE (Sonic Mule) and his presentation including recordings of the Stanford Mobile Phone Orchestra (MoPhO). Become your own iPhone musician by breathing gently into the iPhone’s mic using their Ocarina iPhone application (warning: small fee $0.99).
• Most outstanding announcements: three offerings that open up new user experience opportunities in a Voice 2.0 world (Yes, I already blogged about these but they still stand out as winners for both the service providers and end users.)
  • Voxeo: Tropo.com opens up the Voxeo platform,  previously limited to the small population of voice developers, to hundreds of thousands of web developers.
  • Skype: making the superwideband SILK codec available royalty-free to any relevant hardware or software vendor that wants to bring new user experiences to voice calling.
  • Jaduka: the appointment of Thomas Howe as CEO will accelerate the adoption of voice into existing business processes, eliminating the error-prone mundane while opening up new standards for applications such as healthcare procedures, customer satisfaction and service deployment.
• Other presentations of note:
  • Martin Geddes, Head of Strategy, BT: there is value in addressing issues of efficiency and effectiveness inherent to business processes. Martin proposes that the key business model is to satisfy the needs of parties who want to interact with customers; for instance, improve the productivity of call center employees
  • Shai Berger, Fonolo, whose deep dialing service has now mapped out the IVR trees of over 500 companies and presents new opportunities for customer service operations.
  • Mobivox, expanding their voice-activated mobile services platform with their CRM for Voice business tools.
  • iNum, announcing support for IM and presence, initially with Google Talk
  • Alan Duric continues to see ongoing profitability and success at Norwegian service provider Telio. Why? a continuous stream of innovation and new value-added services.
• Some themes and issues across multiple presentations:
  • Spectrum 2.0: several presentations and a panel covering issues related to (the lack of) spectrum management and “white space”. In major markets over 90% of available spectrum remains unutilized.
  • Regulatory issues: the challenge of creating and executing a user-oriented regulatory framework under “the first technology administration”. Network neutrality: is it more a “buzz phrase” than a real issue? Seeding the policy makers with players who understand regulatory issues from both an innovation and business opportunity orientation as opposed to supporting monopolies or duopolies. Aligning regulation with the application potential of emerging communications technology.
  • API’s: several presentations from vendors who provide free API’s that support third party developers in writing applications that are then hosted by the vendor: IfByPhone, Voxeo, Jaduka, Ribbit, Adhearsion (whose Jay Phillips, right, provided a tutorial incorporating programming to Skype for Asterisk),

One note: there were a lot of innovators and forward thinking people amongst both the speakers and the audience. Most interesting was that there were at least six attendees from carriers who were asking the questions about how carriers can adapt to this emerging communications world: SaskTel (2), T-Mobile US, Rogers, NTT. eComm’s biggest challenge for next year will be to attract a larger audience of carrier personnel. In personal conversations I learned a lot about their issues that may work its way into future commentary.

Overall, eComm 2009 once again provided a primer in a wide range of aspects of the emerging communications space. With Thomas Howe leading Jaduka’s CEBP charge into larger enterprises, Voxeo addressing a huge community of web developers, Skype offering new levels of call quality to the entire communications hardware and software community, Fonolo solving that PITA known as punching your way through company IVR systems and Calliflower offering new user experiences in conference calls, the next few months will prove most interesting. Most challenging to observe will be how these technologies and services assist business and personal issues in times of a most challenging overall economy.

Timothy Hay, Wall Street Journal wrote up Jon Arnold’s panel on “New Ways to Monetize Voice” : VoIP Start-Ups Scramble To Find Niche (subscription may be required)

Alec Saunders: eComm – strong stuff

At last fall’s AstriCon, Stefan Oberg, Skype’s Vice-President for Business, announced the launch of a development program with Digium leading to Skype for Asterisk. Today Digium posted a progress update outlining the current status of the beta program along with more details of the Skype for Asterisk feature set.

Digium claims to have logged “tens of thousands of hours of Skype-to-Asterisk communication”. They’ve also learned a lot about “the art of connecting Asterisk to with the Skype global network”. The post then goes on to provide more details of the forthcoming offering:

The SFA product will be the only solution that integrates Asterisk directly with Skype. This is not a “proxy” solution and the call quality will be superior to anything else on the market. Customers will have the ability to make, receive and transfer Skype calls from within Asterisk phone systems, using existing hardware and existing Asterisk configurations: Skype calls become just another Asterisk call.

Some of the features that will be supported in the market release are:

1. SkypeIn: Receive calls from the public telephone network using standard phone numbers

2. SkypeOut: Make calls to landline and mobile numbers at incredibly low rates

3. Standard phone features: Incoming/outgoing digits (DTMF), Caller ID

4. Smart call routing based on called Skype Name, Caller ID, country of the caller, language they have chosen in their Skype client and etc.

5. Retrieve Skype credit balance information

6. Store and call PSTN and Skype contacts

7. Retrieve and set Skype user presence information

8. Support for G.711 and G.729 voice codecs

9. Each Skype channel license includes a Digium G.729 codec license

Accomplishments to date have been with a very small pilot group of testers; this week they opened up testing to a second larger group. The next stage will be a public beta but no timetable is provided. What I can say is that I have been requested by a beta tester earlier this week to be added as a contact on their Skype for Asterisk test platform.

One interesting statement: “This is not a “proxy” solution and the call quality will be superior to anything else on the market.” Yet the feature set only shows support for narrowband audio G.711 and G.729 codecs. Will there be support of Skype’s SILK super wideband audio for calls routed through Skype for Asterisk that are effectively Skype-to-Skype calls once the end points are connected?

With over 3,000 applications for participation in the public beta, it looks like there is great interest; however, the final proof comes only with a successful launch.

Hat tip to Dan York for his Twitter retweet.

Update: Jason Goecke reports: Experience with the Skype for Asterisk Beta.

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• Skype for Asterisk (stateless.geek.nz)

Following up on Thursday’s announcement of Skype for Asterisk for which details were posted on Skype Journal late Thursday, Dan York has published a more technical post, “Clarifying how Asterisk could possibly be used as a Skype-to-SIP gateway“, discussing how the Asterisk PBX treats incoming and outgoing calls, effectively independently of how the Asterisk PBX is accessed either internally or externally or is directed to make outbound calls.

However, the point I was making in my post yesterday was this announcement has the potential to turn Asterisk into a two-way “Skype-to-SIP” gateway. Asterisk – with the “Skype For Asterisk” module installed – could be deployed into a network where it could provide interconnection between Skype users and SIP users.

Dan goes on in three sections (with diagrams):

• Asterisk Interconnection Explained
• Diving a Bit Deeper
• So How *Might* This Work with ‘Skype for Asterisk”

And, in a concluding section “So What About ‘Skype-to-SIP’ states:

The point of my post yesterday was now that two-way Skype connectivity becomes just another channel driver for Asterisk, you have all sorts of interconnection possibilities. As a standalone system, you could connect SIP phones on an Asterisk server out to the Skype cloud.

If you’re into learning more technical detail of how Asterisk handles and directs inbound and outbound calling, Dan’s post is an excellent primer.

Also check out Dan’s previous post “More on how ‘Skype for Asterisk” actually works…” where he quotes an update post from Tom Keating and concludes:

If I understand this correctly, this has the potential to be huge! As far as I know, all the existing “Skype-to-PBX” solutions use the rather kludgey solution of basically running multiple instances of the Skype client on the system. Each “Skype trunk” is essentially just a separate instance of the Skype client. As Stefan Öberg indicates, there are serious scaling issues with this approach.

However, this has been the only options developers have had! Skype has not – prior to this (if it works how it sounds like it works) – provided any “back-end API” that would let a system interact directly with the Skype P2P cloud. The only API developers have had is the client API that lets them interact with a local Skype client. So that’s how all the “Skype-to-X” products have been built.

Does this mean that Skype has exposed some additional API that is available through this Skype For Asterisk product? If so, this could be VERY interesting…

Interesting times ahead.

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Phil has already pointed out the Skype for Asterisk news announced at Stefan Oberg’s AstriCon keynote this morning along with links to several blog postings and the news release. This afternoon I spent fifteen minutes talking with Stefan, Skype’s Vice-President and General Manager of Telecom, and Digium CEO Danny Windham to get more details.

First I asked who would benefit from the Skype for Asterisk announcement?

Danny and Stefan responded that the primary beneficiary would be the end user, especially small-to-medium businesses who have installed an Asterisk PBX. In particular:

• A generic SkypeID, say “acmesales”, could be setup for inbound calls to the PBX; think of this SkypeID as a “global 800 number”.
• The Asterisk PBX would then be able to hand off the call, as appropriate, to a call center, voice mail, IVR, conferencing and click-to-call from a website, amongst other Asterisk-based services and functionality.
• Each employee or agent can also access the PBX via individual SkypeID’s for taking inbound calls (including calls directed from the generic SkypeID) or placing outbound calls.
• Outbound calls can be placed to any location worldwide, either to a Skype destination or, via SkypeOut, to the PSTN in any country.
• Outbound calls can be to customers anywhere worldwide
• Also the PBX with its Skype inbound/outbound call handling can serve to provide internal company communications amongst offices worldwide. Remote employees are simply at “extensions” of the Asterisk PBX.
• As with any VoIP-based service, agents can be located in remote offices, work from home or be available in any location where they have set up a Skype-enabled PC with broadband access.
• Asterisk PBX already can be programmed to handle least cost routing of international calls; the Skype cloud will be added as an option for least cost routing.
• Calls that involve Skype at both end points will have the full HD (wideband) audio bandwidth of Skype, providing clearer, more readily understood calls than those that involve a PSTN connection at one end.

Naturally the major benefit to end users is the cost savings; Skype to Skype calls are free; calls involving SkypeOut have the normal SkypeOut charges as low as US$ 0.021 or €0.017 per minute. (On SquawkBox this morning Jim Kohlen of the VON Coalition estimated full implementation of VoIP throughout the U.S. could result in savings of up to $110B per year.)

I then probed about the extent of Asterisk installations. It turns out that there were over 1 million downloads of Asterisk via Digium last year; this year is on a run rate of over 1.5 million downloads. Danny estimates there are over 4 million active Asterisk servers worldwide that have been implemented and/or supported by Digium’s various services. Since Asterisk itself is open source, it is speculated there are many more installations out there that are not supported through Digium.

Product: Skype for Asterisk will involve a software module, developed in conjunction with Skype, that is downloaded and compiled onto an Asterisk server. Premium packages will also be available from Skype; these will be comprehensive packages tailored for various business functions and include the Skype Business Control Panel. There may be opportunities to include Skype Partner products and services, such as Pamela and/or PamFax. There will be “low” monthly licensing fees for use of the basic software module as well as the premium packages.

Distribution: Here is where this agreement is significant for Skype. Digium has an established ecosystem involving a market place, technology partners and Value Added Reseller partners (VAR’s). For the VAR’s Skype for Asterisk will be an incremental Digium reseller offering (channel driver) for which they will receive commissions for both the software licenses and premium packages described above as well as for all SkypeOut traffic brought through their customer bases. These VAR’s are responsible for implementation services as well as providing first level technical support to individual customers using Digium products and services.

The Beta program will involve two phases. Phase I will involve a limited number of participants to finalize the software while obtaining feedback from user experiences. Phase II will be a much broader public beta to provide both extended feedback as well as to train VAR’s and even end users on implementation and use of Skype for Asterisk. The beta program will require the use of version 1.4 or 1.6 of Asterisk; Skype for Asterisk will only support these versions once the commercial version is available.

As Rich Tehrani stated in his post:

What this means to Skype is that company has finally found a way to get into the enterprise in an easy way — by partnering with Digium/Asterisk which has great traction with developers, resellers, carriers, SMBs and more. Expect more enterprise use of Skype and as this happens, Skype should see more revenue from business users.

And to narrow down on Dan York’s speculation about any Skype-to-SIP gateway:

• Any existing SIP interfacing functionality within the Asterisk PBX will be available as appropriate to reach non-Skype extensions involving a SIP interface.
• The only additional Skype-to-SIP functionality will come through the existing SkypeOut gateways.

To follow on from my comments yesterday about the need for business transactions related to crossing a SIP interface, both these SIP interfaces will associate with existing business agreements.

Skype has been hinting at major announcements during the fall; this certainly has to be a significant new revenue channel for Skype while bringing new services to Asterisk end users and new sales opportunities for Asterisk resllers.