[OpenSIPS-Users] Incorrect tls port

Sergey Pisanko serp87 at yandex.ru
Wed Jan 5 08:48:09 UTC 2022


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>:

> 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.
>
>
>
>
>
>
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> вт, 4 янв. 2022 г. в 11:47, Bogdan-Andrei Iancu <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
>> 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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>>
>> пн, 3 янв. 2022 г. в 17:59, Bogdan-Andrei Iancu <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
>>> 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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