[OpenSIPS-Users] SIP message relay order

[Bogdan-Andrei Iancu]

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 Developer
http://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 <mailto: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
>     <http://www.opensips.org/html/docs/modules/2.2.x/dialog.html#id297400>
>
>     Regards,
>
>     Bogdan-Andrei Iancu
>     OpenSIPS Founder and Developer
>     http://www.opensips-solutions.com <http://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 <tel:%28514%29%20284-2020>
>>
>>     Email: s <http://firstname.lastname>tas.kobzar at modulis.ca
>>
>>     https://www.modulis.com <https://www.modulis.com/>
>>
>>     _______________________________________________
>>     Users mailing list
>>     Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>     <http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
>
> -- 
>
> 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 <https://www.modulis.com/>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20170207/0891e6ce/attachment-0001.html>