<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></div><br><div id="mt-signature">
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</td><td><span style="color:transparent;font-size:0">01/05/22, 10:45:28 AM</span></td>
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<div>
<font face="monospace">Sergey,<br>
<br>
I see OpenSIPS sents to Asterisk in INVITE:<br>
<br>
Record-Route:
<a><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><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>
<div><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">вт, 4 янв. 2022 г. в 11:47,
Bogdan-Andrei Iancu <<a href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>>:<br>
</div>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<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 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
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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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