[OpenSIPS-Users] SIP message relay order

[Stas Kobzar]

Hello Bogdan and Razvan,

Thank you for sharing your ideas.

Dropping re-INVITE packet to force retransmission sounds really smart
solution for that issue! Also it is easy to implement.
I will give it a try.

Thanks again!

On Sun, Feb 5, 2017 at 11:15 AM, Bogdan-Andrei Iancu <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 <(514)%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

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

https://www.modulis.com
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