Skype Rates and Least Cost Routing

Guest post by Jason Goecke, Adhearsion

Now that Skype is coming to the enterprise with Skype for Asterisk and Skype for SIP, they will need to enhance the data available for their calling rates. Enabling Least Cost Routing (LCR) is a must for any VoIP provider to the enterprise. LCR allows a phone system to determine, on a call by call basis, which VoIP provider to use based on the best rates associated to the country code or prefix being dialed.

As of now Skype publishes a web page of calling rates based on the country name and the per minute rate including or excluding the tax. A few additional items are needed to make this usable for LCR systems:

• The associated country code for each country (i.e. - ‘34′ for Spain, ‘1′ for the US, etc)
• More granular prefixes where calling rates may differ (i.e. - ‘346′ for Spanish mobiles, ‘336′ for French mobiles, ‘1212′ for NYC, ‘1712′ for Iowa, etc)
• Billing intervals
• A file download in CSV, or similar format, for import into LCR systems

Of course, in the meantime it is easy enough to scrape the website and convert the available data into a more appropriate format. Here is an example, in Ruby, of how this may be done in a trivial way:

1. require 'rubygems'
2. require 'open-uri'
3. require 'nokogiri'
4. require 'json'
6. skype_rates = Hash.new
7. skype_url = 'http://www.skype.com/prices/callrates/#allRatesTab'
8. skype_htmldoc = Nokogiri::Hpricot(open(skype_url).read) 
9. (skype_htmldoc/'table.listing//tr.r1').each do |country| 
10.   country_name = country.at('td').inner_html 
11.   skype_rates.merge!({ country_name => { 'amount' => country.at('span.amount').inner_html.split('<!')[0].gsub('$ ', '').to_f, 
12. 'vat' => country.at('span.vat').inner_html.split('<!')[0].gsub('$ ', '').to_f } }) 
13. end
15. p skype_rates.to_json 

Which produces JSON output as follows:

1. { 
2. "Bolivia-La Paz": { 
3. "amount":0.122, 
4. "vat":0.14 
5.   }, 
6. "Sweden - Mobile": { 
7. "amount":0.292, 
8. "vat":0.336 
9.   }, 
10. "Hong Kong": { 
11. "amount":0.021, 
12. "vat":0.024 
13.   } 
14. } 

You may then perform a Regular Expression against another data source to derive the appropriate country codes/prefixes and store those in your LCR system. A good example of the additional detail needed is provided by Flowroute.

I have on my list of actions to create an Adhearsion component to provide LCR capabilities for any Adhearsion application. The plan is to support a wide number of VoIP providers and other data inputs as a part of this plug-in.

In the meantime, it will be interesting to see how Skype goes about publishing their rates with additional details and formats for download.

UPDATE @JimCanuck points out it is not just about least cost, but also about quality of termination. Skype has some interesting approaches to call quality. More here.

tags: skype, lcr, cost, routing, asterisk, adhearsion, voip

Call me at +1-510-455-4384, Skype me, follow @skypejournal and @Phil Wolff.
Visit our Skype Journal private roundtable, one of the longest running public Skype chats.

OnState Virtual PBX: Taking On and Managing All Callers

OnState's Virtual PBX incorporates Skype as a contributor to lower cost, yet more productive, small business communications management activities.

A mainstay for communications in any office with at least a few employees is the need to accept incoming calls, determine who is calling, what is their general need and getting the call to the right employee. In the interest of effective and productive customer and supplier relations, calls need an automated way to reach sales personnel, customer care, accounting or technical support or "the boss". Preferably this call management should be handled without any human intervention; over the past twenty or more years this has resulted in the evolution of increasingly effective Private Branch Exchange ("PBX") offerings from telecomm equipment suppliers. And it traditionally required a reasonably demanding capital expense, from $15,000 up.

The basic functions of a PBX include:

• Receiving and answering a call from an external caller
• Offering a menu of options to determine to whom the call should be connected
• Transferring the call in response to answers provided either by entering alphanumeric or dialpad information or by using speech recognition.
• Accepting, recording and managing voice mails if nobody is available to take a call
• Ability to make a subsequent call transfer if deemed necessary

However, IP-based communications technology along with web 2.0 tools provide opportunities for enhancing the PBX to build more productive and effective business processes when it comes to managing relationships with both suppliers and customers. For instance:

• An individual agent portal for overall conversation management
• Intelligent call queuing would permit an "occupied" employee to either put a caller into a queue for answering when available or transfer a caller to another employee with the skills to handle the call
• Chat sessions can be offered as either an alternative or complement to voice conversations
• Building a searchable call archive integrated into an email system such as GMail.
• Making call transfer destinations independent of the recipient's geographical location, whether "in the office", at a home office or out "on the road".
• Reducing and minimizing the costs associated with various PBX services.

Building on its Call Center experience OnState has launched its Virtual PBX which provides all this functionality as well as:

• Receives calls via Skype, SkypeIn as well as Local DID numbers and toll free numbers in over 20 countries
• Call transfer to employees, agents and other designated recipients on their Skype-enabled PC, landline phones or mobile independent of geographical location
• An extension-based agent/employee/representative portal for managing incoming calls, including call waiting notification, queuing and redirection
• Integration into GMail, Zimbra and Salesforce.com

A few weeks ago, OnState CEO Pat Kelly was our guest on a SquawkBox conference call where he provided more details about OnState's Virtual PBX, including a sharable slide show presentation accessible as a Google document.

Bottom line; for as little as $15 per month per seat and no capital investment, small to medium enterprises and organizations as well can establish an enhanced PBX capability to facilitate both more productive business processes as well as more cost effective communications.

Definitely a business model disruptor.

Powered by Qumana

Dan York Clarifies Skype's Role within Asterisk

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...

Finally, Skype for Asterisk was the topic of discussion for about the first 25 minutes of yesterday's SquawkBox. Access the recording here.

Interesting times ahead.

Powered by Qumana

The Skype for Asterisk Story -- Significant Details

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".
• It will also be Skype-accessible via a click-to-call web button.
  
• The Asterisk PBX would then be able to hand off the call, as appropriate, to a call center, voice mail, IVR, a voice conference and call transfer, 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 Kohlenberger, Executive Director 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 an enhanced 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 390 Value Added Reseller partners (VAR's). For the over 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 [the] 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.
And note that for Skype-to-Skype calls through the Asterisk PBX, there are NO SIP interfaces to/from the PSTN involved; otherwise, there would be no support for HD audio on these calls.
In summary, Skype for Asterisk is a software module providing a Skype cloud-to-Asterisk PBX interface, supporting and interconnecting existing Skype and Asterisk services. It simply uses existing gateways but provides no new SIP gateways.

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 resellers.

Powered by Qumana

Skype for Asterisk gateway software announced

Tom Keating reports from Astricon on Stefan Öberg's announcement of Skype For Asterisk, a channel driver that connects the Asterisk call manager software to the Skype cloud. Register for the driver through a Skype for Asterisk beta program form.

Dan York explains the new value add is "two-way connectivity in and out of the Skype cloud." 

Skype is positioning s4a for business, Asterisk VARs as resellers. This is less about a licensing revenue stream than opening Skype up to the millions of calls managed through Asterisk solutions. 

This will kill off competing Skype channel drivers like Chanskype and create competition to Vosky and other Skype-to-PBX system integrators.

More to come...

Stefan Öberg's blog post and the news release:

Digium and Skype Collaborate to Bring Skype to Business Phone Systems

Skype For Asterisk beta program starts today, adding Skype features to Asterisk-based solutions

GLENDALE, Ariz. (AstriCon 2008)—September 25, 2008—Digium®, creator and primary developer of Asterisk®, the leading open source telephony platform, and Skype™, the leading global Internet communications company, today announced the beta version of Skype For Asterisk, which will allow the integration of Skype functionality into Digium’s Asterisk software and enable customers to make, receive and transfer Skype calls from within their Asterisk phone systems.

“Throughout our individual histories, Skype and Asterisk have each disrupted conventional communication methods through innovative, cost-effective solutions,” said Stefan Öberg, vice president and general manager for Skype Telecom and Skype for Business. “We are excited to be working together with Digium to offer small and mid-sized businesses an even more powerful communications solution to conduct business worldwide.”

Specifically, the beta version of Skype For Asterisk is an add-on channel driver module that integrates Skype Internet calling with Asterisk-based telephony products. Skype For Asterisk also complements small and mid-sized business users’ existing services by providing low rates for calling landline and mobile phones around the world.

“Working together with Skype, our goal is to help businesses boost productivity and reap the rewards of feature-rich telephony software, all while saving a substantial amount of money,” said Danny Windham, CEO of Digium, the creator and sponsor of Asterisk. “The Skype For Asterisk beta program is a first step towards adding Skype capabilities to Asterisk-based phone systems and enabling them to reach more than 338 million Skype users.”

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

• Make, receive and transfer Skype calls from within Asterisk phone systems, using existing hardware.
• Complement existing services with low Skype global rates (as low as 2.1US¢ per minute to more than 35 countries worldwide).
• Save money on inbound calling solutions such as free click-to-call from a website, as well as receive inbound calling from the PSTN through Skype’s online numbers.
• Manage Skype calls using Asterisk applications such as call routing, conferencing, phone menus and voicemail.

Following the beta period when the product is released, Skype For Asterisk will be sold and distributed by Digium and its worldwide network of resellers.

Live at AstriCon

Stefan Öberg will provide the first public demonstration of Skype For Asterisk during his keynote address today at AstriCon, the annual Asterisk user and developer conference. AstriCon attendees are also invited to stop in and see a demonstration of Skype For Asterisk at the Skype booth on the expo floor.

tags: skype, asterisk, business, astricon, pbx, channel, chanskype, pika

Follow Phil Wolff on Twitter or FriendFeed or on Skype.

The Bush/McCain economy is good for Skype

The U.S. misery index is up. Unemployment is at a five year high. The US dollar is at a generational low. Home loans are hard to get and usurious if you get them. College is out of reach for millions. Petrol so expensive that people aren't traveling, are rethinking location decisions like where they work and live, how often they visit family, are cutting shopping trips and buying more online.

This is good for Skype adoption in the United States.

Cheap is Skype's gateway drug.

We substitute onlife communication for costly local and long distance travel. Telecommuting, conference calling, and team chats replace hauling your sorry atoms to meetings. 

We reinforce relationships with family and close friends as financial threats loom large. Safety in numbers, strength in tribes, even at a distance.

We look hard at our monthly spending. Compared to PSTN landlines, $5/month for 10,000 minutes in the US & Canada and a SkypeIn number looks like a lifeline. Hundreds of dollars kept in your wallet. Small businesses, also feeling economic pain, are setting up Skype and Vosky PBX-to-Skype gateways to save. Good feelings in bad times can bank loyalty money can't buy.

Will next month's 2008-Q3 numbers support the theory? We'll see.

tags: skype, economic geography, economy, trends, marketing

Follow Phil Wolff on Twitter or FriendFeed or on Skype.