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<font face="monospace">Hi Sergey,<br>
<br>
If Asterisk is the one changing (from 5061 to 48470) the port in
the RR/Route header, that's illegal to do.<br>
<br>
Regards,<br>
</font>
<pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
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<div class="moz-cite-prefix">On 1/5/22 10:48 AM, Sergey Pisanko
wrote:<br>
</div>
<blockquote type="cite"
cite="mid:CANbQ6F+t5pdw17j=UvxJbBhQ-Qv43NM3NY=MSJDwH=SQgcVeUA@mail.gmail.com">
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<div dir="ltr">
<div dir="ltr">Hi, Bogdan.
<div><br>
</div>
<div>Yes, you are right. That's full call's scheme.</div>
<div><br>
</div>
<div>Opensips:48470 Asterisk
(5062)<br>
1 leg ------------------INVITE (RR:5061)------------><br>
<-----------------INVITE---------------------------------
2 leg</div>
<div>2 leg --------------OK (RR:5061)--------------------></div>
<div><--------------------ACK (Route:48470)------------ 2
leg</div>
<div>< -------------------OK (RR: 48470) -----------------
1 leg</div>
<div>1 leg. ACK From UA1 to Asterisk through Opensips
(Route:48470) sent, but dropped.</div>
<div><br>
</div>
<div><br>
</div>
<div>Best Regards,</div>
<div>Sergey Pysanko.</div>
<div><br>
<br>
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<td><span style="color:transparent;font-size:0">01/05/22,
10:45:28 AM</span></td>
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<div dir="ltr" class="gmail_attr">вт, 4 янв. 2022 г. в 20:44,
Bogdan-Andrei Iancu <<a href="mailto:bogdan@opensips.org"
moz-do-not-send="true">bogdan@opensips.org</a>>:<br>
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<div> <font face="monospace">Sergey,<br>
<br>
I see OpenSIPS sents to Asterisk in INVITE:<br>
<br>
Record-Route:
<a moz-do-not-send="true"><sip:Opensips_IP:5061;transport=tls;lr;ftag=d8e0d49a268d4b51aa85b8f79d2dc062></a><br>
<br>
but in the 200 reply from Asterisk back to OpenSIPS I
see:<br>
<br>
Record-Route:
<a moz-do-not-send="true"><sip:Opensips_IP:48470;transport=TLS;lr;ftag=d8e0d49a268d4b51aa85b8f79d2dc062></a><br>
<br>
Is asterisk the once changing the port there ???<br>
<br>
Regards,<br>
</font>
<pre cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
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<div>On 1/4/22 3:11 PM, Sergey Pisanko wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">
<div dir="ltr">Hi, Bogdan.
<div><br>
</div>
<div>Here is my simple scenario description:</div>
<div><br>
</div>
<div>UA1----Opensips----Asterisk ---- Opensips
----UA2</div>
<div><br>
</div>
<div>Transport protocol doesn't change during this
chain and it's tls, if I understand you right.</div>
<div><br>
</div>
<div>I attached SIP capture of the call. As you can
see, there is the dynamic tcp port in the RR hrd
of last reply to client from which Opensips
connected to the Asterisk. Instead of one, to
which UA1 connected to Opensips (5061). As a
result, there is a media session between UAs, but
only for 30 sec, during of which the UA1 tried to
send ACK to the Opensips, but unsuccessfully for
quite clear reason. Is there the resolution how to
realize this scenario without rewriting RR?</div>
<div><br>
</div>
<div>Best Regards,</div>
<div>Sergey Pysanko.</div>
<div><br>
</div>
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<td><span
style="color:transparent;font-size:0px">01/04/22,
01:46:49 PM</span></td>
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<div class="gmail_quote">
<div dir="ltr" class="gmail_attr">вт, 4 янв. 2022 г.
в 11:47, Bogdan-Andrei Iancu <<a
href="mailto:bogdan@opensips.org"
target="_blank" moz-do-not-send="true">bogdan@opensips.org</a>>:<br>
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<div> <font face="monospace">Hi Sergey,<br>
<br>
Manually altering the RR hdr is a receipt for
disaster :). Somehow I suspect you do not do
double RR (as the protocol changes for the
call). This double RR is automatically done
(by default) when doing `record_route()`. Do
you get 2 RR hdrs when routing the initial
INVITE ?<br>
<br>
Regards,<br>
</font>
<pre cols="72">Bogdan-Andrei Iancu
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<div>On 1/4/22 11:27 AM, Sergey Pisanko wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">
<div dir="ltr">Hello, Bogdan, .<br>
<br>
Thank you for your answer. I've solved my
issue recently just rewriting Record -
Route header with appropriate port within
"onreply route block" by subst function.<br>
<br>
Best Regards,<br>
Sergey Pysanko.<br>
<br>
<br>
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<td><span
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11:27:07 AM</span></td>
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<div dir="ltr" class="gmail_attr">пн, 3
янв. 2022 г. в 17:59, Bogdan-Andrei
Iancu <<a
href="mailto:bogdan@opensips.org"
target="_blank" moz-do-not-send="true">bogdan@opensips.org</a>>:<br>
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<div> <font face="monospace">Hello
Sergey,<br>
<br>
Could you provide a SIP capture (and
calling scenario) to underline the
issue you have ?<br>
<br>
Best regards,<br>
</font>
<pre cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
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<div>On 12/30/21 2:50 PM, Sergey
Pisanko wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">Hello!<br>
<br>
I try to realize the next scenario
with UAs, Opensips-2.4 and
Asterisk.<br>
UAs are registered onto Asterisk
through Opensips and also - on
Opensips if the 200 OK is came
back from Asterisk.<br>
Calls between UAs are relayed to
Asterisk by Opensips.<br>
This scenario works fine with udp.
But it needs to do with tls. And
here I have the problem. What
happens.<br>
Unlike udp, tcp cannot listen its
port and create clients connection
at the same time. Opensips listens
tls port for clients connection <br>
whereas it creates dynamic tcp
port to connect to Asterisk. As a
result, I see that port in
Record-Route header in 200 OK
addressed to caller.<br>
Thus, callers ACK comes to that
dynamic port instead of Opensips
listened port and Opensips dropped
it.<br>
And question is how to force
Opensips to put right port for
caller?
<div><br>
</div>
<div>Regards,</div>
<div>Serhii Pysanko.<br>
<br>
<br>
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<td><span
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