Sergey,
I see OpenSIPS sents to Asterisk in INVITE:
Record-Route:
<sip:Opensips_IP:5061;transport=tls;lr;ftag=d8e0d49a268d4b51aa85b8f79d2dc062>
but in the 200 reply from Asterisk back to OpenSIPS I see:
Record-Route:
<sip:Opensips_IP:48470;transport=TLS;lr;ftag=d8e0d49a268d4b51aa85b8f79d2dc062>
Is asterisk the once changing the port there ???
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
OpenSIPS eBootcamp 2021
https://opensips.org/training/OpenSIPS_eBootcamp_2021/
On 1/4/22 3:11 PM, Sergey Pisanko wrote:
Hi, Bogdan.
Here is my simple scenario description:
UA1----Opensips----Asterisk ---- Opensips ----UA2
Transport protocol doesn't change during this chain and it's tls, if I
understand you right.
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?
Best Regards,
Sergey Pysanko.
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01/04/22, 01:46:49 PM
вт, 4 янв. 2022 г. в 11:47, Bogdan-Andrei Iancu <[email protected]
<mailto:[email protected]>>:
Hi Sergey,
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 ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com <https://www.opensips-solutions.com>
OpenSIPS eBootcamp 2021
https://opensips.org/training/OpenSIPS_eBootcamp_2021/
<https://opensips.org/training/OpenSIPS_eBootcamp_2021/>
On 1/4/22 11:27 AM, Sergey Pisanko wrote:
Hello, Bogdan, .
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.
Best Regards,
Sergey Pysanko.
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01/04/22, 11:27:07 AM
пн, 3 янв. 2022 г. в 17:59, Bogdan-Andrei Iancu
<[email protected] <mailto:[email protected]>>:
Hello Sergey,
Could you provide a SIP capture (and calling scenario) to
underline the issue you have ?
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
<https://www.opensips-solutions.com>
OpenSIPS eBootcamp 2021
https://opensips.org/training/OpenSIPS_eBootcamp_2021/
<https://opensips.org/training/OpenSIPS_eBootcamp_2021/>
On 12/30/21 2:50 PM, Sergey Pisanko wrote:
Hello!
I try to realize the next scenario with UAs, Opensips-2.4
and Asterisk.
UAs are registered onto Asterisk through Opensips and also -
on Opensips if the 200 OK is came back from Asterisk.
Calls between UAs are relayed to Asterisk by Opensips.
This scenario works fine with udp. But it needs to do with
tls. And here I have the problem. What happens.
Unlike udp, tcp cannot listen its port and create clients
connection at the same time. Opensips listens tls port for
clients connection
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.
Thus, callers ACK comes to that dynamic port instead of
Opensips listened port and Opensips dropped it.
And question is how to force Opensips to put right port for
caller?
Regards,
Serhii Pysanko.
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12/30/21, 02:49:47 PM
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