Following a link from Dan York’s update to his post outlining technical details of Skype for SIP, I found a link to this post from IfByPhone’s Irv Shapiro: Why Skype for Asterisk is more important than Skype for SIP.
First, Irv has done perhaps the best articulation yet differentiating Skype for Asterisk from Skype for SIP:
Skype for Asterisk, which is still in closed beta, is a true Asterisk channel driver. This allows Asterisk based solutions to make, receive and transfer Skype calls. A significant capability of the SFA solution is its support for terminating a call to a Skype user name, for example a PC based user of the Skype client.
Skype for SIP is a very different animal. This service provides VOIP trunk support for existing SIP based PBX systems, which may include Asterisk. Unlike SFA where calls may be placed to any Skype user, SFS calls may only be terminated to PSTN end points.
From our perspective as a cloud telephony company, providing hosted telephone applications, SFA is much more interesting. While either service would allow us to provide IVR services to Skype users, the SFA Asterisk channel driver architecture allows us to terminate calls into call centers with no PSTN transport. Each call center Agent would just utilize a headset connected to a computer running the traditional Skype application. Customers calls would be able to originate from either a PSTN device or a Skype client, then route through the Ifbyphone IVR infrastructure and terminate to a call center via Skype’s computer to computer transport. This has the potential to change the cost structure associated with supporting call centers. A Skype based call center would not require a PBX or for even any centralized telephony components. The call center agents could be virtually located anywhere in the world on high quality Internet connections.
Keep in mind that calls terminated on Skype, as demonstrated on the 3 Skypehone service, have no associated termination charges. Long time Skype-based call center provider OnState has already demonstrated this business model; their hosted OnState Call Center service eliminates the need for a call center PBX. While a (prospective) customer can call in via several modes (Skype Online number, DID numbers, 800 numbers, Skype or click-to-call from a website), all the calls are terminated on Skype via their agent client.
Andy Abramson at VoIP Watch has, in a reverse manner, demonstrated this cost advantage also in his post, All Google Voice Needs Is a Little More SIP and Skype’s Game Changes, where he points out that, if Skype for SIP did terminate calls on a Skype client (i.e. – supported SIP—>Skype), then one could terminate Google Voice calls at a Skype client. But, as I have commented on that post and included in Skype for SIP: Sorting Through the Issues, Skype for SIP supports three types of connections and “SIP—>Skype” is not one of them.
Irv demonstrates that Skype for Asterisk has some interesting potential; it’s simply going to require more creative innovation that goes beyond simply making call connections through a SIP interface. It’s a matter of invoking Alec Saunders Voice 2.0 Manifesto which points out that the value lies in the applications.