[OpenSIPS-Users] SIP message relay order

Stas Kobzar stas.kobzar at modulis.ca
Tue Feb 7 08:45:59 EST 2017


Hi Bogdan,

It seems like $DLG_status will not work in my case.

But $DLG_lifetime probably can work. 5 seconds might be too long in my
case, though.
I have re-INVITE almost immediately after initial INVITE (it is Asterisk,
trying to renegotiate SDP media IP/port for direct media).

I guess 1 second might work fine here.

Thank you for your help!


On Tue, Feb 7, 2017 at 5:10 AM, Bogdan-Andrei Iancu <bogdan at opensips.org>
wrote:

> Hi Stas,
>
> If the ACK git the loose_route() / match_dialog(), then the state will
> move to 4.
>
> You may also use the $DLG_lifetime when handling the re-INVITE - if it is
> less than 5 seconds, drop the re-INVITE :)
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 02/06/2017 09:46 PM, Stas Kobzar wrote:
>
> Hello Bogdan,
>
> In my case, ACK for previous INVITE has already been received by OpenSIPS,
> but not sent yet.
> In this case, will the variable $DLG_status still equals 3 ?
>
> Thanks
>
> On Sun, Feb 5, 2017 at 11:15 AM, Bogdan-Andrei Iancu <
> <bogdan at opensips.org>bogdan at opensips.org> wrote:
>
>> Hi Stas,
>>
>> Such races may happen at application level or even at network level (when
>> using UDP) - so if you have 2 packets very close as time, they may swap.
>> That is SIP :)
>>
>> The full guilt is in the UAC device, IMHO - it should let some time gap
>> between the ACK and re-INVITE, to eliminate any possible races.
>>
>> Now, what you can do is to use the dialog module and to check the dialog
>> state when receiving the re-invite. If $DLG_status is *3* (Confirmed by
>> a final reply but no ACK received yet), drop with no reply the re-INVITEs
>> (to force a later retransmission) :
>>     http://www.opensips.org/html/docs/modules/2.2.x/dialog.html#id297400
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>> On 02/02/2017 10:31 PM, Stas Kobzar wrote:
>>
>> Hello List,
>> My call flow has initial INVITE and re-INVITE to update RTP IP/port.
>> Usually everything works well, but sometimes OpenSIPS come up with
>> following example:
>> UA                 OpenSIPS          PSTN GW
>> -------------------------------------------
>> INV(CSeq: 100) -----> | ---> INV(CSeq: 100)
>> <---- 200 OK          | <--- 200 OK
>> (UA sends ACK then new INVITE)
>> ACK(CSeq: 100) -----> |
>> reINV(Cseq: 101) ---> |
>> (OpenSIPS relays first INVITE then ACK)
>>                       | ---> reINV(CSeq: 101)
>>                       | --->   ACK(CSeq: 100)
>> When PSTN gateway receives re-INVITE before ACK for previous INVITE
>> it responds 500 with Retry-After header.
>> This is correct behaviour which conforms to the RFC 3261 section 14.2
>> My question is:
>> Is it possible to assure order of received and relayed messages within
>> the same SIP session? Is there any configuration parameter?
>>
>> Thank you,
>> --
>>
>> Stas Kobzar
>>
>> Developeur VoIP / VoIP Developer
>>
>> Modulis­.ca Inc.
>>
>> # Bureau / Office: 514-284-2020 x 246 <%28514%29%20284-2020>
>>
>> Email: s <http://firstname.lastname>tas.kobzar at modulis.ca
>>
>> https://www.modulis.com
>>
>> _______________________________________________
>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>> --
>
> Stas Kobzar
>
> Developeur VoIP / VoIP Developer
>
> Modulis­.ca Inc.
>
> # Bureau / Office: 514-284-2020 x 246 <(514)%20284-2020>
>
> Email: s <http://firstname.lastname>tas.kobzar at modulis.ca
>
> https://www.modulis.com
>
>


-- 

Stas Kobzar

Developeur VoIP / VoIP Developer


Modulis­.ca Inc.

# Bureau / Office: 514-284-2020 x 246

Email: s <http://firstname.lastname>tas.kobzar at modulis.ca

https://www.modulis.com
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