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