Hi Sergey,
If Asterisk is the one changing (from 5061 to 48470) the port in the
RR/Route header, that's illegal to do.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
OpenSIPS eBootcamp 2021
https://opensips.org/training/OpenSIPS_eBootcamp_2021/
On 1/5/22 10:48 AM, Sergey Pisanko wrote:
Hi, Bogdan.
Yes, you are right. That's full call's scheme.
Opensips:48470 Asterisk (5062)
1 leg ------------------INVITE (RR:5061)------------>
<-----------------INVITE--------------------------------- 2 leg
2 leg --------------OK (RR:5061)-------------------->
<--------------------ACK (Route:48470)------------ 2 leg
< -------------------OK (RR: 48470) ----------------- 1 leg
1 leg. ACK From UA1 to Asterisk through Opensips (Route:48470) sent,
but dropped.
Best Regards,
Sergey Pysanko.
Mailtrack
<https://mailtrack.io?utm_source=gmail&utm_medium=signature&utm_campaign=signaturevirality11&;>
Sender notified by
Mailtrack
<https://mailtrack.io?utm_source=gmail&utm_medium=signature&utm_campaign=signaturevirality11&;>
01/05/22, 10:45:28 AM
вт, 4 янв. 2022 г. в 20:44, Bogdan-Andrei Iancu <[email protected]
<mailto:[email protected]>>:
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 <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 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.
Mailtrack
<https://mailtrack.io?utm_source=gmail&utm_medium=signature&utm_campaign=signaturevirality11&;>
Sender notified by
Mailtrack
<https://mailtrack.io?utm_source=gmail&utm_medium=signature&utm_campaign=signaturevirality11&;>
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.
Mailtrack
<https://mailtrack.io?utm_source=gmail&utm_medium=signature&utm_campaign=signaturevirality11&;>
Sender notified by
Mailtrack
<https://mailtrack.io?utm_source=gmail&utm_medium=signature&utm_campaign=signaturevirality11&;>
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.
Mailtrack
<https://mailtrack.io?utm_source=gmail&utm_medium=signature&utm_campaign=signaturevirality11&;>
Sender notified by
Mailtrack
<https://mailtrack.io?utm_source=gmail&utm_medium=signature&utm_campaign=signaturevirality11&;>
12/30/21, 02:49:47 PM
_______________________________________________
Users mailing list
[email protected] <mailto:[email protected]>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
<http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
_______________________________________________
Users mailing list
[email protected] <mailto:[email protected]>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
<http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
_______________________________________________
Users mailing list
[email protected] <mailto:[email protected]>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
<http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
_______________________________________________
Users mailing list
[email protected] <mailto:[email protected]>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
<http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
_______________________________________________
Users mailing list
[email protected]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
_______________________________________________
Users mailing list
[email protected]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
