<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><div><br></div><div><br><br><br><br><div id="mt-signature">
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</td><td><span style="color:transparent;font-size:0">01/04/22, 01:46:49 PM</span></td>
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<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 style="color:transparent;font-size:0px">01/04/22,
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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">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
<a href="https://www.opensips-solutions.com" target="_blank">https://www.opensips-solutions.com</a>
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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 style="color:transparent;font-size:0px">12/30/21,
02:49:47 PM</span></td>
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