Hi Nick,
It is risky to do a sync sleep as you keep the process blocked, so the
ability of OpenSIPS to handle new messages will decrease .
Either do async sleep, either simple drop it to force a retransmissions
in 500ms - much simpler :)
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 02/07/2017 12:22 PM, Nick Altmann wrote:
The easies way we use in such cases is:
if (is_method("INVITE")) usleep("500");
for in-dialog INVITEs (inside loose_route() / match_dialog() section).
It will delay slightly all reINVITES, but also will guarantee correct
order of packets.
2017-02-06 22:46 GMT+03:00 Stas Kobzar <[email protected]
<mailto:[email protected]>>:
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
<[email protected] <mailto:[email protected]>> 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>[email protected]
https://www.modulis.com <https://www.modulis.com/>
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--
Stas Kobzar
Developeur VoIP / VoIP Developer
Modulis.ca Inc.
# Bureau / Office: 514-284-2020 x 246
Email: s <http://firstname.lastname>[email protected]
https://www.modulis.com <https://www.modulis.com/>
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