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 Developer
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
Email: s <http://firstname.lastname>tas.kob...@modulis.ca
https://www.modulis.com <https://www.modulis.com/>
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