Now that I think about it more, I see one reason why Skype's SIP trunking might be better: the codec.
The codec for SIP/Skype calls is the same idea as codecs for music files: mp3, ogg, wma, etc. You take a drop in quality in exchange for less data. And if you convert from one to another, you take another drop in quality, because each codec strips out different things.
Any of the current solutions (SipToSis, OpenSky, etc) work by taking the output from Skype, converting it to PCM, and converting it to the codec of your choice. This works, but involves a drop in quality. Unlike music, you don't really care if you lose the sound quality of the lead guitarist, because it's a phone call. But if you're a stickler, the drop in quality may bother you.
When you sign up for Skype's beta, they specifically require you to be able to handle the G.729 codec (a common SIP codec). This means one of two things: either Skype is extremely lazy (I haven't ruled that out), or they have some efficient method of converting to and from G.729 and their own proprietary codec -- without converting to PCM, and without a large quality drop. It's possible, because they hold the keys. If that's the case, Skype for SIP or their (eventual) Asterisk channel driver may be worth it for you, if sound quality is a concern. I'd still say give SipToSis a try though.