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<tt>Hi Ben,<br>
<br>
<tt>Sorry to disagree, but IMHO they do exclude one each<tt>
other. <br>
<br>
<tt><tt>Adding RR to TH should not be seen as a way of <tt>fixing
some <tt>broken TH scenarios (w<tt>ith advertise).<br>
<tt>So<tt>,<tt> let me try to understand what is not
<tt>working for y<tt>ou.<tt> You do TH and
adver<tt>tise. In this case, nor<tt>mally,
in the Cont<tt>act headers gener<tt>ated
by OpenSIPS (as a result of TH),
it should be the TH interface<tt>,
right ? What e<tt>xactly seems
to be the problem ? do you
have a trace to show the
issues ?</tt></tt></tt></tt></tt></tt></tt><br>
</tt></tt></tt></tt></tt></tt></tt></tt></tt></tt><br>
<tt>Regards,</tt><br>
</tt></tt></tt>
<pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a></pre>
<div class="moz-cite-prefix">On 27.07.2016 16:05, Newlin, Ben wrote:<br>
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<p class="MsoNormal"><span style="font-size:11.0pt">I understand
that normally you would not need RR with TH, but the two
concepts are not mutually exclusive in SIP. As I said,
</span><span style="font-size:11.0pt">I have a need to
Record-Route the call on my server as I am advertising a
different address than I am listening on. This means that TH
will populate the Contact header with the advertised address
and if I cannot Record-Route with the actual address then I
will not receive sequential requests.</span><span
style="font-size:11.0pt"><o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt"><o:p> </o:p></span></p>
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<p class="MsoNormal"><span
style="font-size:10.5pt;color:black"><o:p> </o:p></span></p>
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</div>
<p class="MsoNormal"><span style="font-size:10.5pt;color:black">Ben
Newlin</span><span style="font-size:11.0pt"><o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt"><o:p> </o:p></span></p>
<div style="border:none;border-top:solid #B5C4DF
1.0pt;padding:3.0pt 0in 0in 0in">
<p class="MsoNormal"><b><span style="color:black">From: </span></b><span
style="color:black">Bogdan-Andrei Iancu
<a class="moz-txt-link-rfc2396E" href="mailto:bogdan@opensips.org"><bogdan@opensips.org></a><br>
<b>Date: </b>Wednesday, July 27, 2016 at 3:59 AM<br>
<b>To: </b>OpenSIPS users mailling list
<a class="moz-txt-link-rfc2396E" href="mailto:users@lists.opensips.org"><users@lists.opensips.org></a>, "Newlin, Ben"
<a class="moz-txt-link-rfc2396E" href="mailto:Ben.Newlin@inin.com"><Ben.Newlin@inin.com></a><br>
<b>Subject: </b>Re: [OpenSIPS-Users] Record-Route and
Dialog topology_hiding()<o:p></o:p></span></p>
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<div>
<p class="MsoNormal"><tt><span style="font-size:10.0pt">Hi
Ben,</span></tt><span
style="font-size:10.0pt;font-family:"Courier
New""><br>
<br>
<tt>As I mentioned in different thread, TH is not
compatible with the RR mechanism. If you do TH, your
OpenSIPS will act as and end point (from SIP
perspective), so there will be no Route/RR headers at
all. So no need to do loose_route or so. You just do
TH matching for the sequential requests and nothing
more.</tt><br>
<br>
<tt>Regards,</tt><br>
<br>
</span><o:p></o:p></p>
<pre>Bogdan-Andrei Iancu<o:p></o:p></pre>
<pre>OpenSIPS Founder and Developer<o:p></o:p></pre>
<pre><a moz-do-not-send="true" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a><o:p></o:p></pre>
<div>
<p class="MsoNormal">On 22.07.2016 16:48, Newlin, Ben
wrote:<o:p></o:p></p>
</div>
<blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
<p class="MsoNormal"><span style="font-size:11.0pt">Hi,</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"> </span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt">I am
using the Dialog module with topology_hiding() in my
server and I have a need to Record-Route the call on
my server as I am advertising a different address than
I am listening on. I have found what I believe is an
inconsistency in the handling of Record-Route within
the Dialog topology_hiding functionality. The
topology_hiding isn’t a true B2BUA, but it does set up
different parameters for the incoming UAC and outgoing
UAS sides of the call for the Via headers,
Record-Route and Route headers, and the Contact
header(s).</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"> </span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt">The
problem is that the record_route() and loose_route()
functions operate on different sides of the call. The
record_route() function will only add a Record-Route
header to the outgoing UAS side of the call. And since
the record_route() function cannot be called from
onreply_route, but is no way to add a Record-Route
header to the UAC side of the call.</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"> </span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt">On the
other hand, the loose_route() function only operates
on the incoming UAC side of the call and there is no
way to perform loose_route() on the UAS side of the
call.</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"> </span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt">So
there is a situation where Record-Route headers can
only be added on the outgoing UAS side, but the
associated Route headers can only be removed on the
incoming UAC side (where they won’t exist since they
can’t be added) and any added headers on the UAS side
cannot be processed properly due to the lack of
loose_route.</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"> </span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt">I can
provide further information if this is unclear. It
should be easily reproducible by attempting to use
record_route in a topology_hiding scenario. The route
is added to the outbound leg, but is not removed by
loose_route so the message is looped back every time.</span><o:p></o:p></p>
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style="line-height:16.8pt;background:white"><b><span
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style="font-size:10.0pt;font-family:Arial;color:#666666"> </span></b><span
style="font-size:10.0pt;font-family:Arial;color:#666666">| Sr Voice
Network Engineer, PureCloud</span><o:p></o:p></p>
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