Re: [OpenSIPS-Users] Incorrect tls port

[Sergey Pisanko]

Bogdan, thanks a lot for your replies!

Best Regards,
Sergey Pysanko.

On Wed, Jan 5, 2022, 16:37 Bogdan-Andrei Iancu  wrote:

> I mean, as per SIP, the UAS device must mirror, without any changes, the
> received RR into the 200 OK replies. And here even if Asterisk receives the
> RR hdr with the 5061 port, it sends back a 200 OK with a 48470 port in RR
> :-/
>
> 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 4:32 PM, Sergey Pisanko wrote:
>
> Bogdan.
>
> Is it refers to the specific Asterisk behaivior scheme below? Asterisk's
> ACK of leg 2 and 200 OK of leg1 must be addressed to Opensips port 5061?
>
> Best Regards,
> Sergey Pysanko.
>
> On Wed, Jan 5, 2022, 15:54 Bogdan-Andrei Iancu 
> wrote:
>
>> 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)>
>> > < ---OK (RR: 48470) - 1 leg
>> 1 leg. ACK From UA1 to Asterisk through Opensips (Route:48470) sent, but
>> dropped.
>>
>>
>> Best Regards,
>> Sergey Pysanko.
>>
>>
>>
>> [image: Mailtrack]
>> 
>>  Sender
>> notified by
>> Mailtrack
>> 
>>  01/05/22,
>> 10:45:28 AM
>>
>> вт, 4 янв. 2022 г. в 20:44, Bogdan-Andrei Iancu :
>>
>>> Sergey,
>>>
>>> I see OpenSIPS sents to Asterisk in INVITE:
>>>
>>> Record-Route:
>>> 
>>>
>>> but in the 200 reply from Asterisk back to OpenSIPS I see:
>>>
>>> Record-Route:
>>> 
>>>
>>> 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:
>>>
>>> UA1OpensipsAsterisk  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.
>>>
>>>
>>>
>>>
>>>
>>>
>>> [image: Mailtrack]
>>> 
>>>  Sender
>>> notified by
>>> Mailtrack
>>> 
>>>  01/04/22,
>>> 01:46:49 PM
>>>
>>> вт, 4 янв. 2022 г. в 11:47, Bogdan-Andrei Iancu :
>>>
 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.



 [image: Mailtrack]
 
  Sender
 notified by
 Mailtrack
 
  01/04/22,
 11:27:07 AM

 пн, 3 янв. 2022 г. в 17:59, 

Re: [OpenSIPS-Users] Incorrect tls port

[Bogdan-Andrei Iancu]

I mean, as per SIP, the UAS device must mirror, without any changes, the 
received RR into the 200 OK replies. And here even if Asterisk receives 
the RR hdr with the 5061 port, it sends back a 200 OK with a 48470 port 
in RR :-/


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 4:32 PM, Sergey Pisanko wrote:

Bogdan.

Is it refers to the specific Asterisk behaivior scheme below? 
Asterisk's ACK of leg 2 and 200 OK of leg1 must be addressed to 
Opensips port 5061?


Best Regards,
Sergey Pysanko.

On Wed, Jan 5, 2022, 15:54 Bogdan-Andrei Iancu > wrote:


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)>

Sender notified by
Mailtrack


01/05/22, 10:45:28 AM   


вт, 4 янв. 2022 г. в 20:44, Bogdan-Andrei Iancu
mailto:bog...@opensips.org>>:

Sergey,

I see OpenSIPS sents to Asterisk in INVITE:

Record-Route:



but in the 200 reply from Asterisk back to OpenSIPS I see:

Record-Route:



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:

UA1OpensipsAsterisk  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


Sender notified by
Mailtrack


01/04/22, 01:46:49 PM   


вт, 4 янв. 2022 г. в 11:47, Bogdan-Andrei Iancu
mailto:bog...@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 

Re: [OpenSIPS-Users] Incorrect tls port

[Bogdan-Andrei Iancu]

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)>
 
	Sender notified by
Mailtrack 
 
	01/05/22, 10:45:28 AM 	



вт, 4 янв. 2022 г. в 20:44, Bogdan-Andrei Iancu >:


Sergey,

I see OpenSIPS sents to Asterisk in INVITE:

Record-Route:



but in the 200 reply from Asterisk back to OpenSIPS I see:

Record-Route:



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:

UA1OpensipsAsterisk  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


Sender notified by
Mailtrack


01/04/22, 01:46:49 PM   


вт, 4 янв. 2022 г. в 11:47, Bogdan-Andrei Iancu
mailto:bog...@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.



Mailtrack


Sender notified by
Mailtrack


01/04/22, 11:27:07 AM   


пн, 3 янв. 2022 г. в 17:59, Bogdan-Andrei Iancu
mailto:bog...@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 

Re: [OpenSIPS-Users] Incorrect tls port

[Sergey Pisanko]

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)>

Sender
notified by
Mailtrack

01/05/22,
10:45:28 AM

вт, 4 янв. 2022 г. в 20:44, Bogdan-Andrei Iancu :

> Sergey,
>
> I see OpenSIPS sents to Asterisk in INVITE:
>
> Record-Route:
> 
>
> but in the 200 reply from Asterisk back to OpenSIPS I see:
>
> Record-Route:
> 
>
> 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:
>
> UA1OpensipsAsterisk  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.
>
>
>
>
>
>
> [image: Mailtrack]
> 
>  Sender
> notified by
> Mailtrack
> 
>  01/04/22,
> 01:46:49 PM
>
> вт, 4 янв. 2022 г. в 11:47, Bogdan-Andrei Iancu :
>
>> 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.
>>
>>
>>
>> [image: Mailtrack]
>> 
>>  Sender
>> notified by
>> Mailtrack
>> 
>>  01/04/22,
>> 11:27:07 AM
>>
>> пн, 3 янв. 2022 г. в 17:59, Bogdan-Andrei Iancu :
>>
>>> 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.
>>>
>>>
>>>
>>> [image: Mailtrack]
>>> 
>>>  Sender
>>> notified by
>>> Mailtrack
>>> 

Re: [OpenSIPS-Users] Incorrect tls port

[Bogdan-Andrei Iancu]

Sergey,

I see OpenSIPS sents to Asterisk in INVITE:

Record-Route: 



but in the 200 reply from Asterisk back to OpenSIPS I see:

Record-Route: 



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:

UA1OpensipsAsterisk  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 
 
	Sender notified by
Mailtrack 
 
	01/04/22, 01:46:49 PM 	



вт, 4 янв. 2022 г. в 11:47, Bogdan-Andrei Iancu >:


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.



Mailtrack


Sender notified by
Mailtrack


01/04/22, 11:27:07 AM   


пн, 3 янв. 2022 г. в 17:59, Bogdan-Andrei Iancu
mailto:bog...@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.



Mailtrack


Sender notified by
Mailtrack


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


___
Users mailing list
Users@lists.opensips.org  
http://lists.opensips.org/cgi-bin/mailman/listinfo/users  





___
Users mailing list
Users@lists.opensips.org 

Re: [OpenSIPS-Users] Incorrect tls port

[Sergey Pisanko]

Hi, Bogdan.

Here is my simple scenario description:

UA1OpensipsAsterisk  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.






[image: Mailtrack]

Sender
notified by
Mailtrack

01/04/22,
01:46:49 PM

вт, 4 янв. 2022 г. в 11:47, Bogdan-Andrei Iancu :

> 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.
>
>
>
> [image: Mailtrack]
> 
>  Sender
> notified by
> Mailtrack
> 
>  01/04/22,
> 11:27:07 AM
>
> пн, 3 янв. 2022 г. в 17:59, Bogdan-Andrei Iancu :
>
>> 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.
>>
>>
>>
>> [image: Mailtrack]
>> 
>>  Sender
>> notified by
>> Mailtrack
>> 
>>  12/30/21,
>> 02:49:47 PM
>>
>> ___
>> Users mailing 
>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

--
UA1 to OPENSIPS Opensips_IP:5061:
--

INVITE sip:508080@asterisk_IP:5062;transport=tls SIP/2.0
Via: SIP/2.0/TLS 
UA1_IP:52732;rport;branch=z9hG4bKPjb0cc2bffdc81454eb26583c7d3925484;alias
Max-Forwards: 70
From: ;tag=d8e0d49a268d4b51aa85b8f79d2dc062
To: 
Contact: 
Call-ID: adb91a11aeb84cebbb1a0e9b3f737f9c
CSeq: 9147 INVITE
Route: 
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, 
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: MicroSIP/3.20.5
Authorization: Digest username="505050", realm="asterisk", 
nonce="1641298342/d7e3fb8b3d36162e02c6f06ba7ce025d", 
uri="sip:508080@asterisk_IP:5062;transport=tls", 
response="ef89c414b398b86f836c647a4618716e", algorithm=md5, 
cnonce="882c1c44e3634a188fee3c3cfcea4893", opaque="6a1b7faf68185146", qop=auth, 
nc=0001
Content-Type: application/sdp
Content-Length:   693

v=0
o=- 3850294342 3850294342 IN IP4 UA1_IP

Re: [OpenSIPS-Users] Incorrect tls port

[Bogdan-Andrei Iancu]

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.



Mailtrack 
 
	Sender notified by
Mailtrack 
 
	01/04/22, 11:27:07 AM 	



пн, 3 янв. 2022 г. в 17:59, Bogdan-Andrei Iancu >:


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.



Mailtrack


Sender notified by
Mailtrack


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


___
Users mailing list
Users@lists.opensips.org  
http://lists.opensips.org/cgi-bin/mailman/listinfo/users  





___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Incorrect tls port

[Bogdan-Andrei Iancu]

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.



Mailtrack 
 
	Sender notified by
Mailtrack 
 
	12/30/21, 02:49:47 PM 	



___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] Incorrect tls port

[Sergey Pisanko]

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.



[image: Mailtrack]

Sender
notified by
Mailtrack

12/30/21,
02:49:47 PM
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users