Switching from Skype to SIP, and Asterisk PBX

I’ve been using Skype for a while, and it mostly works OK, though I wish they’d hurry up and release a Linux client that uses ALSA for sound directly, instead of requiring OSS emulation.

The Skype conferencing feature works well as long as everyone has a reliable connection. But it’s irritating that if someone drops out, and needs to rejoin, that the conference initiator has to reinvite them. I use the conference a lot when friends and I play Ticket to Ride online, and one of them has a lot of network problems, so this is a real hassle.
I’m considering switching to using SIP, which is not a product but rather an open VOIP protocol. Most VOIP products other than Skype support SIP, so they can all interoperate, whereas Skype doesn’t interoperate with any other service. And there are many SIP-based hardware products, such as phones, ATAs, and routers with integral ATAs, while there are only a few hardware products that work with Skype.
There are both commercial and free VOIP clients for Windows, Linux, MacOS, and other platforms. I just compiled Ekiga 2.00 Beta 1 on Linux, and it seems to work fine. Ekiga is the new name of the program that used to be called GnomeMeeting. They changed the name to emphasize that it isn’t just a clone of NetMeeting, but is in fact a full SIP and H.323 client. I’ve only used Ekiga for a short time, but it seems to work fine.

I’ve compiled the Asterisk PBX softtware on my server, in order to try using it as a conference bridge, but I haven’t yet learned how to configure it.

For Windows, it looks like Gizmo (free, but not open source) and OpenWengo (GPL) should work fine as SIP clients, but there are many others. Google has indicated that Google Talk will support SIP in the future; right now it only supports XMPP.

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