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 48470 port
in RR :-/
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 4:32 PM, Sergey Pisanko wrote:
Bogdan.
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?
Best Regards,
Sergey Pysanko.
On Wed, Jan 5, 2022, 15:54 Bogdan-Andrei Iancu <[email protected]
<mailto:[email protected]>> wrote:
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 <https://www.opensips-solutions.com>
OpenSIPS eBootcamp 2021
https://opensips.org/training/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.
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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.
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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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