This morning Skype and Digium announced the release of Skype for Asterisk, an Asterisk platform channel driver that allows businesses using Asterisk to incorporate Skype into their calling options.

Since first announcing the Skype for Asterisk beta program last fall, over the past eleven months the beta program has gradually expanded its scope to a broader range of Asterisk developers. The result is that not only has Digium been leading the program involving over 4,000 external participants but also they have incorporated Skype for Asterisk into their own communications operations with Skype accounts such as “Digium Sales” and “Digium Support”, typical of the convention they expect many businesses to follow.

The Skype for Asterisk launch press release includes identification of a couple of Digium Partners providing implementation of Skype for Asterisk to their clients as well as some features and pricing information:

DATUS Corporation is a Digium Select Partner with nearly four decades of experience designing and implementing communications networks in Germany. The company has nearly completed an Asterisk installation at 2,100 sites for LVM Versicherungen, a major insurance firm, and also works with Digium to design features for Asterisk that are of particular interest to European businesses. “Adding Skype for Asterisk to the DATUS indali OBX, our IP-PBX, will offer our customers inexpensive and secure international calling that, for instance, could be used for toll free customer services,” said Jonny Kueppers, vice president of sales and marketing at DATUS. “We believe that the price and cost savings will be welcome with today’s budgets.”

“The combination of Skype and Asterisk gives those companies that have relied on Skype the advanced call management capabilities of Asterisk, while Asterisk users get free calling to more han 400 million registered Skype users and low Skype rates when calling landlines and mobiles,” said Stefan Öberg, vice president and general manager of Skype for Business. “We believe the product will bring together two of the largest groups of users that value flexibility and cost savings in their PBX systems.”

Foehn Ltd, Digium’s U.K. Solutions Partner, has been designing and implementing Asterisk-based solutions for more than five years. The company’s technical director, James Passingham, commented: “With Skype for Asterisk, we can offer our clients even more freedom in business communications. The ability to unlock the lower call costs of Skype provides a huge savings opportunity, especially for those with offices and customers around the globe.”

Skype for Asterisk Features

Skype for Asterisk, which is compatible with the free and open source Asterisk versions 1.4, 1.6 and AsteriskNOW™, as well as the commercially licensed Asterisk  Business Edition™, is unique in the market today. It is the only solution that integrates directly with Skype, enables multiple concurrent Skype calls from a single Skype account, and supports both G.711 and G.729a calling.

In addition, with the new software, customers can:

• Manage business Skype accounts with the Business Control Panel
• Get low Skype global rates on outbound calls (as low as 2.1US¢ per minute to more than 35 countries worldwide; connection fee and fair usage policies apply)
• Receive inbound calls to online numbers
• Route calls according to Skype profile fields, online status and privacy settings
• Streamline customer contact by allowing Web site visitors to place free Skype calls directly to their business with global click-to-call buttons

Pricing and Availability

Skype for Asterisk is available to download now for $66 USD per concurrent call from http://www.digium.com/skype or from Digium Authorized Resellers and Distributors worldwide. Skype for Asterisk comes with 90 days of installation support from the time of purchase.

More details, including documentation and a Buy button, are at Digium’s Skype for Asterisk product page. Update: Fierce VoIP, Skype for Asterisk now available, includes comments from Digium’s Pete Engler and Skype’s Matthew Jordan.

Within the next couple of weeks I will be discussing an implementation of Skype for Asterisk that I have been beta testing for the past few weeks.

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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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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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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.