[inquisitor]

Originally Posted by hrgajek

I tested the SIP feature with my Speedport W700V Router (delivered by Deutsche Telekom)

That device is probably not the best choice for a test. I would rather try an AVM FritzBox, which usually delivers best user experience for VoIP.

I received calls only when calling my US-number, then the mobile phone kept quiet. (Mobile phone was registered in German E-Plus radio-network) When I switched off the VoIP-Account in the router, the mobile phone rang again.

It seems like they don't support parallel call (mobile and SIP ringing simultaneously), which I found a great feature with solomo, who however never solved the billing issue of their parallel call feature while roaming and eventually dismissed it.

I could not make outgoing calls via SIP. I didn't find out, which area codes are necessare, as I do not know, how and where the SIP-Server is connected to the PSTN, if he is. (Liechtenstein, Germany, US, anywhere?)

I guess their SIP infrastructure is hooked up to Mobilkom Liechtenstein's core network and so SIP calls are originating from Liechtenstein.

To be honest: I'm not so familiar with VoIP-Things, so I do not know, if I did all setups correctly.

If the phone rang for incoming calls and your VoIP router was not attached to another NAT-router there's few you could have mistaken with configuration. But I wouldn't rule out that there was a problem with audio codecs between the SIP server and your router.

But lets keep realistic. How should their business modell work, if callers could reach the mobile via SIP for free?

I didn't propose free incoming calls on the mobile, but rather the following two features:

1. Receive calls by SIP originating from a SIP-party (no radio network involved): If you're being called on one of the DID numbers or directly by VoIP through a SIP-URI you could answer calls by a VoIP client, which would mean for FTT (or their underlying MNO-partners) they would take the termination fees (at least for the German and Liechtenstein number, where the calling-party-pays scheme applies) but there would be no costs for terminating the call except for the negligible costs of IP transit of the SIP traffic. Inbound pure SIP calls (from a SIP party through SIP-URI answered by SIP client) would of course mean no revenue, but the costs of this is again just IP transit.
   
2. Receive calls on the mobile from a SIP-party (using radio network): If you're called on your SIP-URI but answer it on your mobile (as opposed to your SIP client as above) FTT would of course need to charge their customers for the incoming call. This feature of course would rather target a small fraction of their customers who are VoIP-savvy.

I think, to answer all this questions, you should invest in your own SIM-card

My readiness to invest time and money in beta tests has declined over the past years. I've helped United Mobile to discover some anti-roaming-steering tricks and USSD-blocking in Bulgaria and despite my help they didn't even refund the erroneously billed calls back then. Of course if they gave me a SIM for free I would be happy to test it.