[OpenSIPS-Users] Incorrect tls port
Bogdan-Andrei Iancu
bogdan at opensips.org
Tue Jan 4 18:42:31 UTC 2022
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.
>
>
>
>
>
>
> 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 <bogdan at opensips.org
> <mailto:bogdan at opensips.org>>:
>
> 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
>> <bogdan at opensips.org <mailto:bogdan at opensips.org>>:
>>
>> 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
>>> Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users <http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
>>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org <mailto:Users at lists.opensips.org>
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> <http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20220104/d500cba5/attachment-0001.html>
More information about the Users
mailing list