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<font face="monospace">I mean, as per SIP, the UAS device must
mirror, without any changes, the received RR into the 200 OK
replies. And here even if Asterisk receives the RR hdr with the
5061 port, it sends back a 200 OK with a </font><font
face="monospace"><a rel="noreferrer">48470 port in RR :-/<br>
<br>
Regards,<br>
</a></font>
<pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
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<div class="moz-cite-prefix">On 1/5/22 4:32 PM, Sergey Pisanko
wrote:<br>
</div>
<blockquote type="cite"
cite="mid:CANbQ6FJ9FJ_mqUOFPRR4zmRj0bUnfe8KA+Adyz+NhHdxNBmimQ@mail.gmail.com">
<meta http-equiv="content-type" content="text/html; charset=UTF-8">
<div dir="auto">Bogdan.
<div dir="auto"><br>
</div>
<div dir="auto">Is it refers to the specific Asterisk behaivior
scheme below? Asterisk's ACK of leg 2 and 200 OK of leg1 must
be addressed to Opensips port 5061?<br>
</div>
<div dir="auto"><br>
</div>
<div dir="auto">Best Regards,</div>
<div dir="auto">Sergey Pysanko.</div>
</div>
<br>
<div class="gmail_quote">
<div dir="ltr" class="gmail_attr">On Wed, Jan 5, 2022, 15:54
Bogdan-Andrei Iancu <<a href="mailto:bogdan@opensips.org"
moz-do-not-send="true">bogdan@opensips.org</a>> wrote:<br>
</div>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">
<div> <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 cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
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<div>On 1/5/22 10:48 AM, Sergey Pisanko wrote:<br>
</div>
<blockquote type="cite">
<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
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10:45:28 AM</span></td>
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<div class="gmail_quote">
<div dir="ltr" class="gmail_attr">вт, 4 янв. 2022 г. в
20:44, Bogdan-Andrei Iancu <<a
href="mailto:bogdan@opensips.org" target="_blank"
rel="noreferrer" 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 rel="noreferrer"
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 rel="noreferrer"
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
<a href="https://www.opensips-solutions.com" target="_blank" rel="noreferrer" moz-do-not-send="true">https://www.opensips-solutions.com</a>
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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>
</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 dir="ltr" class="gmail_attr">вт, 4 янв.
2022 г. в 11:47, Bogdan-Andrei Iancu <<a
href="mailto:bogdan@opensips.org"
target="_blank" rel="noreferrer"
moz-do-not-send="true">bogdan@opensips.org</a>>:<br>
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style="margin:0px 0px 0px
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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
OpenSIPS Founder and Developer
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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>
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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"
rel="noreferrer"
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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