<div dir="auto">Did you add the handle_ruri_alias() as suggested by Daniel? I had something like this where I would get “unable to resolve blah blah blah" and it’s because the RURI is the actual wss “address” which is unresolvable, so executing the function forces kamailio to take the alias instead.</div><div dir="auto"><br></div><div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Fri, 7 May 2021 at 13:48, Igor Olhovskiy <<a href="mailto:igorolhovskiy@gmail.com">igorolhovskiy@gmail.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div>
<p>Daniel,</p>
<p>Seems to be it's really the case, but with other function</p>
<p>With FQDN in RR <br>
</p>
<h3> <code>is_first_hop()</code></h3>
<p>is not acting correctly for reply. <br>
</p>
<p>
</p><blockquote type="cite">For incoming SIP replies, it means that
top Record-Route URI is 'myself' and source address is not
matching it</blockquote>
But in Record-Route we have "myself", but <b>is_first_hop()</b>
returning 0. <br>
<p></p>
<p>Thanks!<br>
</p>
<pre cols="72">Regards,
Igor</pre></div><div>
<div>On 07.05.2021 14:22, Daniel-Constantin
Mierla wrote:<br>
</div>
<blockquote type="cite">
<p>OK, because looping was something that should not have happened
in this case.</p>
<p>Then the problem is that you do not do nat-traversal-like
processing for websocket/webrtc traffic. You have to use
set_contact_alias() + handle_ruri_alias() because the webrtc
endpoints do not set "valid" contact addresses.</p>
<p>Cheers,<br>
Daniel<br>
</p>
<div>On 07.05.21 14:13, Igor Olhovskiy
wrote:<br>
</div>
<blockquote type="cite">
<p>Ah, no, sorry, I was wrong at this one,</p>
<p>This just is not sent with "unable to resolve address
toleivu2gdbh.invalid". <br>
</p>
<p>Sorry. Looping were something else during my tests, this just
with <b>advertise</b> added<br>
</p>
<pre cols="72">Regards,
Igor</pre>
<div>On 07.05.2021 14:02,
Daniel-Constantin Mierla wrote:<br>
</div>
<blockquote type="cite">
<p>This looks like incoming ACK, because there is only one Via
header, so it is not what proxy forwards -- that one is
relevant to see what headers were consumed and added.</p>
<p>Cheers,<br>
Daniel<br>
</p>
<div>On 07.05.21 13:51, Igor Olhovskiy
wrote:<br>
</div>
<blockquote type="cite">
Sure.
<p><font face="Courier New, Courier, monospace">ACK <a>sip:88290@toleivu2gdbh.invalid;transport=wss</a>
SIP/2.0<br>
Via: SIP/2.0/UDP
A_IP_ADDRESS:5060;rport;branch=z9hG4bKPj8d05548a-91ef-4332-8617-32f8eeebf8f2<br>
From: <a><sip:88881@A_IP_ADDRESS></a>;tag=0a3e31a6-96ad-42d0-9310-81b35cedbd3d<br>
To: <a><sip:88290@KAMAILIO_FQDN></a>;tag=hvra7mj3q0<br>
Call-ID: 46f44741-d155-4dd5-8fd8-78e540fc1acb<br>
CSeq: 18326 ACK<br>
Route: <a><sip:KAMAILIO_FQDN;lr;r2=on;ftag=0a3e31a6-96ad-42d0-9310-81b35cedbd3d;nat=wss></a><br>
Route: <a><sip:KAMAILIO_FQDN:8089;transport=ws;lr;r2=on;ftag=0a3e31a6-96ad-42d0-9310-81b35cedbd3d;nat=wss></a><br>
Max-Forwards: 70<br>
User-Agent: Asterisk PBX 13.33.0<br>
Content-Length: 0</font></p>
<p><br>
</p>
<p>By loop I meant, Kamailio just relaying it back to self
and discard.<br>
</p>
<pre cols="72">Regards,
Igor</pre>
<div>On 07.05.2021 13:48,
Daniel-Constantin Mierla wrote:<br>
</div>
<blockquote type="cite">
<p>Can you paste the ACK that loops, after being handled
once by Kamailio?</p>
<p>Cheers,<br>
Daniel<br>
</p>
<div>On 07.05.21 13:25, Igor
Olhovskiy wrote:<br>
</div>
<blockquote type="cite">
<p>Daniel,</p>
<p>Yes, it is.</p>
<p><font face="Courier New, Courier, monospace">alias=...<br>
</font></p>
<p>Also tried with</p>
<p><font face="Courier New, Courier, monospace">listen =
IP advertise FQDN</font><br>
</p>
<p>same behavior, <font face="Courier New, Courier,
monospace">loose_route()</font> stops acting
correctly.<br>
</p>
<p>PS: Forgot to add, Kamailio 5.4.3 / 5.4.4<br>
</p>
<pre cols="72">Regards,
Igor</pre>
<div>On 07.05.2021 13:21,
Daniel-Constantin Mierla wrote:<br>
</div>
<blockquote type="cite">
<p>Hello,</p>
<p>is the KAMAILIO_FQDN set as local domain for
Kamailio (via alias parameter or domain
module+register myself)?</p>
<p>Cheers,<br>
Daniel<br>
</p>
<div>On 07.05.21 11:17, Igor
Olhovskiy wrote:<br>
</div>
<blockquote type="cite">
<p>Hello,</p>
<p>I saw there are some topics on this already and
carefully walked through all of them, but can't
solve following issue.</p>
<p>For a reason I do need that in Record-Route
header sent to endpoint would present FQDN. If it
matters, it's UDP/WSS conversion done on Kamailio.</p>
<p>So, scheme is quite simple<br>
</p>
<p>Enpoint A ->UDP-> Kamailio ->WSS->
Endpoint B (NAT)<br>
</p>
<p>Main issue here, that if in Record-Route headers
it's FQDN, but not IP addresses, on a new
transactions with a dialog (ACK on 200, PRACK,
BYE), Kamailio<b> loose_route()</b> function
resolves address of destination not current
dialog, but actual R-URI (or itself, if R-URI is
something from WebRTC world) that is not correct
due to NAT.<br>
</p>
<p>If in RR headers IP addresses are present - all
is working as expected.</p>
<p>As an example (RR with FQDN)<br>
</p>
<p>B answers 200</p>
<p><font face="Courier New, Courier, monospace">SIP/2.0
200 OK<br>
Record-Route: <a><sip:KAMAILIO_FQDN:8089;transport=ws;r2=on;lr=on;ftag=0a3e31a6-96ad-42d0-9310-81b35cedbd3d;nat=wss></a><br>
Record-Route: <a><sip:KAMAILIO_FQDN;r2=on;lr=on;ftag=0a3e31a6-96ad-42d0-9310-81b35cedbd3d;nat=wss></a><br>
Via: SIP/2.0/UDP
<A_IP_ADDRESS>:5060;received=A IP
ADDRESS;rport=5060;branch=z9hG4bKPj67fb6d86-97d7-4231-995b-e54b0f62881e<br>
To: <<a>sip:88290@</a><KAMAILIO_FQDN>>;tag=hvra7mj3q0<br>
From: <<a>sip:+XXXX7688881@</a><KAMAILIO_FQDN>>;tag=0a3e31a6-96ad-42d0-9310-81b35cedbd3d<br>
Call-ID: 46f44741-d155-4dd5-8fd8-78e540fc1acb<br>
CSeq: 18326 INVITE<br>
Contact: <a><sip:88290@toleivu2gdbh.invalid;transport=wss></a><br>
Allow:
ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER<br>
Supported: outbound<br>
Content-Type: application/sdp<br>
Content-Length: 817</font></p>
<p><br>
</p>
<p>ACK looks like</p>
<p><font face="Courier New, Courier, monospace">ACK
<a>sip:88290@toleivu2gdbh.invalid;transport=wss</a>
SIP/2.0<br>
Via: SIP/2.0/UDP
A_IP_ADDRESS:5060;rport;branch=z9hG4bKPj8d05548a-91ef-4332-8617-32f8eeebf8f2<br>
From: <a><sip:88881@A_IP_ADDRESS></a>;tag=0a3e31a6-96ad-42d0-9310-81b35cedbd3d<br>
To: <a><sip:88290@KAMAILIO_FQDN></a>;tag=hvra7mj3q0<br>
Call-ID: 46f44741-d155-4dd5-8fd8-78e540fc1acb<br>
CSeq: 18326 ACK<br>
Route: <a><sip:FQDN;lr;r2=on;ftag=0a3e31a6-96ad-42d0-9310-81b35cedbd3d;nat=wss></a><br>
Route: <a><sip:FQDN:8089;transport=ws;lr;r2=on;ftag=0a3e31a6-96ad-42d0-9310-81b35cedbd3d;nat=wss></a><br>
Max-Forwards: 70<br>
User-Agent: Asterisk PBX 13.33.0<br>
Content-Length: 0</font><br>
</p>
<p>And Kamailio on <b>loose_route()</b> loops ACK
to itself. (with result of function == 1)<br>
</p>
<p>In a case if in Record-Route/Route headers just
IP address of Kamailio present, all works as
expected, but it's not really behavior I want to
achieve.</p>
<p>I've tried to play with <b>record_route_preset("...")</b>
specifying only WSS part (as suggested in <a href="https://skalatan.de/de/blog/kamailio-sbc-teams" target="_blank">https://skalatan.de/de/blog/kamailio-sbc-teams</a>)
with FQDN, but no luck.</p>
<p>Also wanted to try approach using
record_route_preset() function in branch route,
but it was working only with first branch, not
affecting others (but I assume having different RR
headers across branches is not a good idea)</p>
<pre cols="72">--
Regards,
Igor</pre>
<br>
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</blockquote>
<pre cols="72">--
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<fieldset></fieldset>
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</blockquote>
<pre cols="72">--
Daniel-Constantin Mierla -- <a href="http://www.asipto.com" target="_blank">www.asipto.com</a>
<a href="http://www.twitter.com/miconda" target="_blank">www.twitter.com/miconda</a> -- <a href="http://www.linkedin.com/in/miconda" target="_blank">www.linkedin.com/in/miconda</a>
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</blockquote>
</blockquote>
<pre cols="72">--
Daniel-Constantin Mierla -- <a href="http://www.asipto.com" target="_blank">www.asipto.com</a>
<a href="http://www.twitter.com/miconda" target="_blank">www.twitter.com/miconda</a> -- <a href="http://www.linkedin.com/in/miconda" target="_blank">www.linkedin.com/in/miconda</a>
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</blockquote>
</blockquote>
<pre cols="72">--
Daniel-Constantin Mierla -- <a href="http://www.asipto.com" target="_blank">www.asipto.com</a>
<a href="http://www.twitter.com/miconda" target="_blank">www.twitter.com/miconda</a> -- <a href="http://www.linkedin.com/in/miconda" target="_blank">www.linkedin.com/in/miconda</a>
Kamailio Advanced Training - Online
May 17-20, 2021 (Europe Timezone) - June 7-10, 2021 (America Timezone)
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</blockquote>
</div>
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</blockquote></div></div>-- <br><div dir="ltr" class="gmail_signature" data-smartmail="gmail_signature"><div dir="ltr"><div>Regards,</div><div><br></div>David Villasmil<div>email: <a href="mailto:david.villasmil.work@gmail.com" target="_blank">david.villasmil.work@gmail.com</a></div><div>phone: +34669448337</div></div></div>