SIP is the underlying protocol that makes most VOIP work. If you're using Vonage, or Asterisk, or most other VOIP systems/providers, your phone calls are getting coordinated over SIP, with the audio sent back and forth on the side. Using SIP, Cisco systems can communicate with Asterisk systems, which can communicate with Microsoft SoundPoint systems, etc. Any of those systems can connect to a "SIP Provider" to get phone service.
Skype is off in its own little walled garden, with a special protocol and codec. There have been many attempts to link Skype and SIP, and they're usually pretty painful (and proprietary).
SipToSis is a program that will allow you to have a skype "server" that will connect sip calls to skype users and vice versa. It's a bit of a pain to set up, but it walks. He also offers a set of scripts to have multiple skype clients set up, load and unload them as necessary, redirect calls, etc. It's a huge, huge hack, but it works, and is much cheaper than previous solutions of this type.
There was apparently a beta test for an Skype channel driver for asterisk. This would allow someone to setup skype as just another input type (like a Zaptel analog phone connection, or a SIP trunk), and seemed to be the ideal solution. Either it never went anywhere, or they decided they didn't want me in the beta
:(
Gizmo also offers a Skype trunking solution, similar to what Skype seems to be offering. They call it
OpenSky. It looks like it would work pretty well for home users, but it would get pretty steep for businesses -- and how many home users would set up friggin asterisk, besides me?
So if you're a business, OpenSky or Skype's current beta is probably what you're looking for. If you're a home user or an admin who either can't wait or has too much time on your hands, give SipToSis a try. It's a bit of a pain to set up, but it costs $2-$14 dollars one time, as opposed to everyone else, who will charge monthly or per minute.