One of the major complaints about Skype is it’s a closed system. Skype provides PSTN connectivity (SkypeOut and SkypeIn) but no way to connect to other VoIP services. Of course 3rd parties have been providing work arounds, but until now there’s been nothing from Skype provoking major complaints from some in the community (see this letter from Gizmo Project founder Michael Robertson).
The Skype-Asterisk deal announced this morning means Skype is officially supporting multi-channel connectivity to an open source SIP platform. That in turn means an Asterisk box can function as a gateway between Skype and any other SIP-based system, inlcuding Gizmo. If the Asterisk box is used as a gateway, the Asterisk dial plan adapts between Skype names, SIP URIs and PSTN numbers. It’s early days, but this looks like a significant win for every VoIP community. It’s also likely to provide a boost to Skype’s efforts to attract business users.
They announced an early beta program starting today, to be followed by a public beta and, presumably, a stable release at some point. The additions to Asterisk will be licensed code available from Digium at a price and the Skype-specific code will not be open, but Mark Spencer, CEO of Digium and founder of Asterisk, suggested the connectivity code would be licensed much as G.729 code is today. (G.729 code requires a license, not for the software, but for to cover royalties to patent holders).
We’ll await further details, but this looks to be very significant. Stay tuned.