Here are some details. I haven't tried it yet...<br />
Eventually my curiosity prevailed, so I subscribed to them through someone else's account. Indeed, they're pretty much relying on DeltaThree's existing infrastructure. The SIP server is in the US, in DeltaThree's subnet, and the RTP server (for audio) is at BezeqInt.
UPDATE: This LINK has a very interesting conversation on the topic.
I haven't been able to get the service working for quite some time as I don't have the user/password. If you do get it working drop me a note :)
Eventually my curiosity prevailed, so I subscribed to them through someone else's account. Indeed, they're pretty much relying on DeltaThree's existing infrastructure. The SIP server is in the US, in DeltaThree's subnet, and the RTP server (for audio) is at BezeqInt.
Registrar: 213.137.73.140 (this was found by sniffing - it's not apparent from the Windows client)To login with Ekiga:1. Open Edit | Accounts.2. Add account:Name: BezeqInt (or whatever)
User: 972744141000 (your phone number)<br /><br />Password: (your VoIP password)<br /><br />3. Open "More Options"<br /><br />Authentication Login: (your VoIP user ID -- not the same as your phone number)
Realm/Domain: deltathree.com<br /><br />4. Save the account and mark the checkbox to connect.<br /><br />5. Open Edit | Preferences.<br /><br />6. Go to Protocols | SIP Settings<br /><br />7. Set Outbound Proxy to 213.137.73.140 (or else you wouldn't be able to make outgoing calls, since your origin address won't be @213.137.73.140)
The RTP server only offers 8kbps codecs so it doesn't give better quality than regular phone lines. Also, for some reason, Ekiga seems to reset its outgoing calls (incoming calls are fine) as soon as the remote end answers. It goes:
Me (Ekiga client): SIP INVITE sip:[EMAIL PROTECTED]<br /><br />Server: SIP 100 Trying<br />Server: SIP 180 Session Progress<br />Server: RTP audio messages begin to arrive<br />Server: SIP 200 OK<br />Me: ICMP UDP port unreachable for all RTP audio messages following the OK<br />Server: SIP 200 OK (<-- yes, once more)<br />Me: ICMP UDP port unreachable for the 2nd OK<br /><br />While at it, Ekiga says "Abnormal call termination". I can only wonder what makes Ekiga think things are not kosher and abandon the INVITE.
UPDATE: This LINK has a very interesting conversation on the topic.
I haven't been able to get the service working for quite some time as I don't have the user/password. If you do get it working drop me a note :)
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