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 <stas.kob...@modulis.ca>: > 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 <bog...@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.kob...@modulis.ca >> >> https://www.modulis.com >> >> >> _______________________________________________ >> Users mailing >> listUsers@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.kob...@modulis.ca > > https://www.modulis.com > > _______________________________________________ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >
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