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

[bugzilla-daemon at freedesktop.org]

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

--- Comment #6 from Mikhail Zabaluev <mikhail.zabaluev at nokia.com> 2011-03-22 05:28:02 PDT ---
(In reply to comment #4)
> 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"

Yes, that would be a good clue. Besides, Telepathy-Rakia does not hide URI
parameters, so if you do not see this in the contact ID, it was not in the
INVITE. The display alias attribute (from Aliasing) omits the URI parameters
for readability.

> For all I know it's not THAT easy to fake an inbound INVITE.

I mean sending an invite with a From header of your choice. Standard SIP has no
ways of verifying senders' identity. In fact, in many network configurations
anyone can send a request directly to our UA stack's listening socket.

But all told, there could be an option to interpret "phone-like" user names in
SIP URIs of the same domain as the account owner's; presumably SIP services use
authentication to verify senders claiming to be from their domain. We could use
Addressing to offer tel: interpretation of such URIs.

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