[Bug 35437] decode tel: URIs and expose over D-Bus to allow PSTN heuristics

bugzilla-daemon at freedesktop.org bugzilla-daemon at freedesktop.org
Mon Mar 21 17:57:25 CET 2011


https://bugs.freedesktop.org/show_bug.cgi?id=35437

--- Comment #4 from Joerg Reisenweber <joerg_freedesktop-org-bugzilla at omma-projekt.de> 2011-03-21 09:57:25 PDT ---
(In reply to comment #3)
> (In reply to comment #0)
> > 23:38 < DocScrutinizer> Robot101: there's a problem with telepathy-sofiaSIP,
> > contacts, and dialer. I got an account at sipgate.de, a german SIP provider. It
> > has an associated landline number. Inbound call from e.g landline "+49 911
> > 123456" are signalled and logged as origin:"SIP:0911123456 at sipgate.de".
> 
> If this is how it looks in SIP messages, there is no way to tell it is actually
> a phone number.
This is how it looks when Maemo displays the info it gets from sofiaSIP. The
real SIP INVITE definitely has better details. Please also note my odd worded
comment about 
call=tell that AIUI actually should read ";user=phone"
(http://de.wikipedia.org/wiki/Session_Initiation_Protocol). However take this
with a grain of salt, as I never seen a provider supporting it yet, and I'm not
sure it's even supposed to be used that way.

> Heuristics are prone to guessing wrong
given the fact most SIP providers don't even allow inbound direct SIP via IP,
there's not much more guessing involved than just looking for a settings flag
set for that particular SIP account that tells sofiaSIP about the properties of
this account.

> and getting into bigger
> problems for unsuspecting users (faking a PSTN caller with a simply set up SIP
> proxy, anyone?).
For all I know it's not THAT easy to fake an inbound INVITE. Actually I'd think
a properly implemented SIP stack should ignore such faked INVITE msgs (not to
mention you'll have a hard time to get *any* fake thru a NAT, but that's
unrelated and irrelevant here)

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