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More on how "Skype For Asterisk" actually works...

Does "Skype for Asterisk" tear down some of Skype's walls? (and allow SIP-to-Skype?)

skype_logo.pngdigiumlogo.gifDoes today's announcement of a beta version of "Skype for Asterisk" signal a way to tear down some of Skype's walls? And does this move Skype along toward better SIP interoperability?

The announcement happened out at Astricon today and TMC's Tom Keating had one of the first posts about it - updated with info from TMC reporters who are at Astricon. Both the Digium news release and the Skype blog post highlight these four points that Asterisk users will be able to do:

  • Make, receive and transfer Skype calls with multiple Skype names from within Asterisk phone systems, using existing hardware.
  • Complement existing Asterisk services with low Skype global rates (as low as 1.7€¢ / 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 throughcreate virtual offices all over world using Skype’s online numbers.
  • Manage Skype calls using Asterisk applications such as call routing, conferencing, phone menus and voicemail.

I want to focus on one part of the first bullet. Recall that in my last post about Skype and SIP interoperability I talked about how Skype currently has one-way connectivity via SIP to external SIP clouds. A SIP system can receive calls from a Skype user, but cannot make calls into Skype's cloud. (My employer Voxeo's application platform is one example.) Yet here's the first bullet of the announcement:

  • Make, receive and transfer Skype calls with multiple Skype names from within Asterisk phone systems, using existing hardware.

Ta da... two-way connectivity in and out of the Skype cloud.

What's more, because Asterisk is really a telephony platform that speaks multiple protocols, you could easily see the ability to interconnect into other systems... including SIP clouds. Here's a quick graphic showing how it could work:


I changed the color of the arrows to and from the PSTN to reflect the fact that PSTN connectivity could really occur from either the Skype cloud or directly from the Asterisk system. Conceivably you might have an Asterisk system with existing PSTN connectivity (through either hardware cards or SIP or IAX trunks) that only wants to use the Skype For Asterisk channel driver to communication to/from Skype users. On the other end, Asterisk can connect to systems running the protocols of SIP, H.323 or SCCP (Cisco Skinny), as well as whatever other protocols Asterisk sysadmins might add to their Asterisk box. They could be on-premise systems such as IP-PBXs or they could be hosted systems or "clouds" of network connectivity.

Now what is really being announced today is that you can register to join the beta program for Skype For Asterisk. You cannot download the code yet. You can't inspect it to see how it works. All we can do is speculate and sign up to join the beta program (which, it indicates, may involve an NDA).

Still, it's an interesting move and it will be intriguing to see how this actually works.

I'd like to understand, though, how this is or is not similar to what has been offered by Chanskype for a few years now. Is this the same code? i.e. did Skype buy or obtain the Chanskype code or team? Given the lack of any info in that regard in these announcements, I'm inclined to think it is separate code.... but how does it compare?

We'll have to see as the code becomes available. Tom Keating did say in his post that it will not be available as open source code but rather under a commercial license.

In the meantime, congrats to both the folks at Digium and Skype for making this happen - and I look forward to seeing it in action.

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