[OpenSIPS-Users] Incorrect tls port

Bogdan-Andrei Iancu bogdan at opensips.org
Wed Jan 5 13:54:48 UTC 2022


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.
>
>
>
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> 	01/05/22, 10:45:28 AM 	
>
>
> вт, 4 янв. 2022 г. в 20:44, Bogdan-Andrei Iancu <bogdan at opensips.org 
> <mailto:bogdan at opensips.org>>:
>
>     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
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>>     	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
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>>>         	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
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>>>>             <https://mailtrack.io?utm_source=gmail&utm_medium=signature&utm_campaign=signaturevirality11&>
>>>>             	12/30/21, 02:49:47 PM 	
>>>>
>>>>
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>>>
>>
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