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I would guess that this move is designed to ensure that the cellular "pay by minute" model does not go the way of the dinosaurs. If we are moving towards mobile broadband connections on all cell phones, it's not a far cry to have users install Skype or other generic VoIP clients on their phones to make calls (especially to bypass those expensive international rates).

Also, I would guess that this move allows the carrier to control content and maintain a licensing fee or commission for software placed on their phones. (I guess the Apple iPhone would be an exception...seems like AT&T is bending ov^H^H^H^H^H^H^H^H^H^H not getting a good deal here).

You can do this today, in the US, using the HTC Wizard (aka Cingular 8125) and a SIP-enabled softphone like SJPhone. This app is pretty demanding of the Wizard, but the newer phones (TyTN, Artemis, etc.) should be able to handle it no problem.

Here's a link from 2004 about how to do something similar.

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I saw your post on Ekiga 2.0 when I googled for other stuff, but it was to late to replay and I can't find a function for sending private messages so I'll answer here instead. Sorry for beeing off topic.

Regarding SIP and Ekiga I'm not sure if there are any chances to use SIP behind a NAT without any forwarded ports (in that case using STUN but I doubt it.)
What ports are used are application dependant, but this is what I use and I have no troubles with any application or provider so far:

5060 TCP/UDP (SIP UAS, I think this should be enough.)
5000-5100 UDP (But with only the ports above I could only use my other account, not the Ekiga.net one, with these both works.)
3478-3479 UDP (STUN service)
49152-65535 UDP (RTP, RTCP multimedia streaming)

I _think_ that port 5060 (or 5000-5100) is used to say "hi, you got a phone call", but the actual voice data are sent over port 49152-65535 (valid for SJPhone, might be application dependant). I think you only need the STUN ports open if you use STUN, STUN seems to be a service which helps the application to figure out what kind of firewall it is behind.

I've covered this with screenshots in my blog but it's written in swedish so I don't know how much help that gives. Atleast you can look at the images?
I know for my "real provider" I had to tell them if I where behind NAT or not, doesn't seem like anything like that is needed for the ekiga.net account for whatever reason. Maybe with STUN + NAT enabled user you don't have to forward any ports at all?

Another alternative is to use Asterisk or Asterisk@home somewhere outside the firewall and have that handle the SIP account and use IAX for your clients instead. There is a nice (atleast screenshots says so) client available for UNIX called KIAX.

Ekiga is a nice client, for whatever reason it crashes all the time now when I upgraded to Ubuntu Dapper, but I've got noone else to blame than myself for that one. SJPhone isn't open-source but it doesn't cost anything and is very competent aswell. For Linux linphone is less good and kphone is even less good ;), I would rate X-ten X-lite(is that the name?) as just beneath SJPhone.

Earlier I used an ATA box and a SIP VoIP from Bredbandsbolaget (sweden, all of the rest aswell) and then I switched to Rix Telecom which I only use together with a softphone, I'm considering switching to Affinity Telecom since you don't pay anything per month and get a free ATA box if you sign up for 12 months (I haven't found the catch in that...)

Anyway, I call with software only and yes it works. For Windows and others gizmo project has a client which seems nice, but I think it only works with their servers. SJlabs SJphone is the client I thought where best in Windows and there is a Linux version aswell. Another client is Xten X-lite which I avoided since the GUI looked so weird (it tend to do on them all.)
Free as in speach clients for the UNIX world has been quite crappy in comparision, the best are probably linphone both kphone might work aswell.

Anyway yes it do works! In Linux with ALSA I would rate the sound quality ekiga / SJphone on first place, then linphone and last kphone. For the interface SJphone, ekiga, kphone, linphone. The bad thing with SJphone in Linux was that it didn't had any calling tones if you didn't set up a command yourself to run when you got incoming calls. But I didn't know if it would kill it aswell or repeat it or anything so I never did and therefor never knew if anyone called ;)

Ekiga is good.

I have an Acer N50 PDA with a 500MHz CPU and Skype really does work just fine. So does SJphone (SIP softphone) which can also make pretty good calls although seems more sensitive to weak wireless signal.

Alternatively, if you have a PocketPC PDA, you can run the freeware SIP phone SJPhone on it. Also works great with Asterisk.

How about xten, firefly, pulver, sjphone, sflphone, minisip, kphone, linphone, phonegaim? This is just the software I know of, there are probably many more that other people can add to the list. Which that would suit you depends of course on what your needs are.

I prefer FreeWorld Dialup as a great way to go for voip:
 
- standards based
- Free
- Windows , Linux and pocketpc clients available
- Call 800 numbers and more
- Call to/from vonage customers
- get free phone number and have people call you
- Get a wisip phone (WiFi SIP) and you have the closest thing to a IP mobile phone you can get.
 
  FreeWorld Dialup

FreeWorldDialup has been doing pretty much the same thing for years now. They even have their own pre-configured software. If you don't like the Pulver Communicator you can always get one of half a dozen other SIP phones (soft or otherwise) that work with the service. They even offer a pre-configured version of The SJPhone, which is essentially what the Gizmo folks are offering. If you want a dial-out service FWD offers a choice of affiliates who can give you that portion a-la-carte!

So, why would you want to use Gizmo?

Or, you could get Xten's Softphone, or SJ Labs and use it with your home VOIP.

As someone else mentioned - FreeWorld Dialup is a great way to go for voip:

- standards based
- Free
- Windows, Linux and pocketpc clients available
- Call 800 numbers and more
- Call to/from vonage customers
- get free phone number and have people call you
- Get a wisip phone (WiFi SIP) and you have the closest thing to a IP mobile phone you can get.

FreeWorld Dialup

I'm currently running windows and ipaq (pocketpc 2002) clients fine. And calling my home vonage service - no problem!

VoIP on a PocketPC is a kludge at best for now. Forget about wardriving (warwalking?) with a PDA. The CF and internal WiFi cards are much lower power than laptop PCMCIA cards so you won't have the range you need. Battery life with the WiFi radio running is pretty pitiful too. Not a huge problem with outgoing calls, but forget about leaving it connected in standby to receive calls.

Still, if you want to try it, there's a few SIP softphones for PocketPC, Xten and SJPhone. Never tried them myself, but they're freeware, so doesn't cost anything to experiment.