Notices

got it.... oubound proxy had to be set to 'getonsip.com'

with that get on sip stuff can i call skype? i registerd it seems pretty universal mode of video comms as minimal downloads are necessary

its a good option

I would like to know that myself..... I know next to nothing re these voip alternatives to skype...

Apparently so but it's a little more involved on the callers part.

And GV/Google Talk works similarly

same case with me btw its a nice solution its pretty has minimal fuss but still needs education to use it on every device its not like sign up enter your username pwd and done iwas wondering as the n900 cannot use web rtc we use the inbuilt sip system then say i want to call user x with device y its still a fuss to get him register to sip then setup bla etc web rtc is nice but it isint everywhere atleast its available in dextop browsers mostly whats intresting is one can just send a link and get going in web rtc mode

my question is can we call a dextop browser from the n900 i.e imean the user on the other end would not be needed to fuss with a sip client

Comfytel offer a similar skype service as well.

Interesting. In case of my provider (website in polish only, so I guess not much sense in posting it here) I never required to set up anything from "advanced" field - no idea why username field is there, if you already provide your username in basic settings. But there are reports of it being required for certain SIP proxies?... Same situation with outbound proxy.

BTW, technically, SIP is per-to-peer, and you don't need proxy, at all. you could just have your username@xxx.xxx.xxx.xxx, where xxx part is IP. Or, for more fancy things, set up asterisk on your device (it is in repos).

But, the sip proxies ('providers') give you convenience of a) working SIP no matter which connection you're using (behind NAT/firewall), b) calls to legacy (PSTN) phone numbers (for a price).

Again, technically, you could use your own PBX (asterisk powered, in home, for example), to power up your own VoIP<->PSTN gateway, for times when you need to call analog/cellular numbers. But, for most people, relying on some provider... well, providing that functionality, is more feasible.

So, SIP 'providers' are not about central server - those proxies just translate your IP's from behind NAT. During SIP<->SIP calls, whole conversation is peer-to-peer - audio/video streams aren't forwarded by proxy, just initial SIP call establishing packets. Your conversation streams passes through "provider" infrastructure, when you're using ther VoIP<->PSTN gateway, though, for payed calls to regular numbers.

/Estel

@estel...
I have a linux home server... Is it possible (and easy) to set up one's own sip server? If so can you point me to an idiot's guide on the subject..
SIP to skype would also be very useful...

I am working at the moment on psi/psi+ and psimedia for N900.

I hope I will get it running. If you are interessted there is a experimental binary for psi (psi15.0.maemo5_binary.tar.gz). You should use the menu quit entry to exit the application.
Sound and video is not working atm because psimedia is not finished.
However also psi is only a little bit polished so that you can use the most gui windows/dialogs.

Can't give you LMGTFY link that would reflect up-to-date state of things, as I haven't been through process of setting it up for years - hoever, search for asterisk tutorial, and you'll hit zillions of guides. I suspect that - if anything - it become easier with time, not more complicated. I would start on official asterisk web page:
http://www.asterisk.org/

For sure, it shouldn't be complicated for you if you've already set up your linux server in home and you're used to tinker with configuration files, etc. Furthermore, after short glance at their website, it seems that they've streamlined setting up process, making it all modular.

/Estel