Digium Provides Progress Update on Skype for Asterisk

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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