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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Ran this by users, 1 responder agreed, no work around. Seems
like a bug, but I wanted to get some more feedback before reporting anything.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Asterisk authenticates on invite using this methodology;<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>First, the name portion of the URI on the from header of the
invite is compared to the defined sip users, so if an invite comes from <a
href="mailto:user1@abc.com">user1@abc.com</a> asterisk looks for a sip user
named ‘user1’. If that sip user is found the invite is expected to
pass the configured authentication for that user.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>If there is no match the source IP/port of the invite is
compared to statically configured users (host=, port=, insecure=).<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><br>
The resulting behavior is that there is a bug when a SIP invite coincidentally originates
‘from’ a trusted IP address, but the user in the URI matches a
configured user in the target system. Asterisk rejects the invite with a 407,
proxy authentication required.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>The correct behavior (I think) would be to evaluate possible
matches to statically defined users FIRST (port AND IP for security), then look
at the username if there is no match. If it were not for NAT you would not parse
the URI either, you would look for a complete match of the from URI in the
invite and the registered users ‘FULL CONTACT’ URIs.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Situations were SIP users are trusted by IP address alone
are typically peering situations, and in most conceivable implementations the number
of peering partners is far fewer than the number of registered UAs, but the number
of calls from an individual peering partner is likely to be far greater than the
number of calls from any singe UA. There would be a resulting slight efficiency
gain since the higher number of inbound calls from a trusted peer would find a
match earlier in the evaluation. I realize this is arguable and subject to the
situation at hand, but the point is that it would eliminate a bug and NOT
create any IN-efficiency.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Have I missed a simple concept? Is there a way to send a
user/pass pair in the invite that overrides Asterisk authentication methodology?<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Right now it seems truly impossible to fully trust a user by
IP, since an invite can be formed in such a way that certain calls are rejected
with the SIP 407 message.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>The ultimate question is whether this is a bug or a training
issue.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Damon E<o:p></o:p></span></font></p>
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