Skype for SIP == Skype for Asterisk DOA?

Guest post by Jason Goecke, Adhearsion

Today Skype announced Skype for SIP (SFS). Put simply, enterprise telephone systems may now interconnect with the Skype network to receive calls from the Skype network and place calls to SkypeOut. All without the need to install any special hardware or software on most modern enterprise phone systems (IP-PBXs to be more specific). Skype’s new enterprise targeted connectivity uses SIP, the industry standard for VoIP interconnection. SIP already powers the bulk of Skype’s revenue, via SkypeIn/SkypeOut, so this is a logical progression to take advantage of the large scale infrastructure already in place at Skype.

This is a tremendous move by Skype and one I have contended for years was necessary for them to make headway in the enterprise. I applaud this step. There are plenty of great posts out there covering this already, including the one by @danyork on Disruptive Telephony.

What does this mean for Skype for Asterisk (SFA) announced last September? At best the value of SFA has been significantly reduced by this announcement.

Previously SIP interconnection to the Skype cloud was given to the rarified group of larger players such as Voxeo, Tellme, Genesys and others. SFA was the first time this access was going to be brought to the world of open source telephony developers through Asterisk. This provided an immense opportunity for the Asterisk developer community to create new applications to take advantage of this, which lead me to invest time to participate in the closed beta for SFA still underway.

The SFS announcement this morning has just marginalized SFA to applications that benefit from direct dialing of Skype users from Asterisk and from basic presence updates from the Skype network. Gone are the benefits of providing Skype/SkypeIn inbound calls to the enterprise, SkypeOut trunking, etc. More so, SFA is at a disadvantage since you will have to pay a per channel (simultaneous call) license fee on top of any SkypeIn/SkypeOut costs. Further, I suspect that the number of SFA channels available to a single account will be limited for the same reason that SFS does not do SIP to Skype dialing, so that no one may provide large scale alternatives to SkypeIn.

All of this has really taken the wind out of the SFA sails before it even had a chance to make it to a public beta. Digium must now look to quickly add new features. Such as advanced presence information, instant messaging, the SILK codec and others, if they hope to salvage their own investment in the development of SFA to date. While I understand these things take time, the lethargy of getting the SFA to market does not bode well for rapidly trumping the SFS announcement.

Time will tell.

tags: asterisk, ip-pbx, sfa, sfs, sip, skype, skypeforasterisk, skypeforsip, voiceoverip, VoIP, jasongoecke

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