The Session Initiation Protocol (SIP) has proven to be an effective standard for Voice over IP (VoIP) systems, as it’s use in open source and closed source products is now relatively quickly displacing the H.323-based systems that preceded it. H.323 and its accompanying protocols weren’t necessarily bad, but they come out of the videoconferencing space and brought along a lot of baggage not needed for audio calls. H.323 was also not designed (or at least not well designed) for extensibility to generalized session management, either.
Nearly two years ago Skype showed up on the scene with yet another approach to VoIP. Skype uses a proprietary P2P protocol for making VoIP calls. While the bandwidth usage of Skype is quite high (in some cases requiring even more bandwidth than a PSTN call), it makes up for that weakness by avoiding the most common issues with firewalls, routers, and Network Address Translation (NAT) that afflict most SIP and H.323 based solutions.
NAT is a technology that, among other things, uses private IP addresses in order to hide one or more PCs behind a router and a single routable IP address. Rather than having to forward ports through the firewall, which is a bit much to ask a typical home user to do, Skype tunnels voice packets over either port 80 or 443. Port 80 is generally used for http and port 443 for https. Since these ports are used by web browsers, they are generally left open on a router, at least for requests sent from behind the router and for responses to those requests.
For many SIP-based softphones, you have to “punch” holes in your firewall for a few ports. H.323-based solutions are often even worse. I was able to get Skype to work easily on Windows and Linux without making any changes to the configuration of my router. I did run into a few problems with sound on Linux (as did at least 10 other people who either commented on one of my posts or emailed me), but I recently installed Skype again and it appears to be working fine. I’m wombatnation on Skype, though I’m not often online.
Advanced IP Pipeline reports that the IETF is now considering P2P extensions to SIP. That could solve a lot of the problems that home users still have with SIP-based solutions. There are also rumors that Skype has started to use SIP in its platform. Hopefully they will drop their proprietary protocol entirely. Dealing with several different IM clients was painful enough until Gaim had come far enough along to handle all the different major IM service providers.
References: I found the article on P2P extensions to SIP via SIPThat.com blog. SIPThat is run by the President and co-founder of Xten, makers of the X-Lite SIP softphone of which I’m quite fond (despite the confusing behavior of the preferences dialogs). Another blog I read for info on SIP is Mr. Blog, which is run by a researcher at Earthlink.