[OpenSIPS-Users] Incorrect tls port

Bogdan-Andrei Iancu bogdan at opensips.org
Tue Jan 4 09:45:14 UTC 2022


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
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 	
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>
> 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.
>>
>>
>>
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>>     	12/30/21, 02:49:47 PM 	
>>
>>
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