Bogdan, thanks a lot for your replies! Best Regards, Sergey Pysanko.
On Wed, Jan 5, 2022, 16:37 Bogdan-Andrei Iancu <bog...@opensips.org> wrote: > I mean, as per SIP, the UAS device must mirror, without any changes, the > received RR into the 200 OK replies. And here even if Asterisk receives the > RR hdr with the 5061 port, it sends back a 200 OK with a 48470 port in RR > :-/ > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS eBootcamp 2021 > https://opensips.org/training/OpenSIPS_eBootcamp_2021/ > > On 1/5/22 4:32 PM, Sergey Pisanko wrote: > > Bogdan. > > Is it refers to the specific Asterisk behaivior scheme below? Asterisk's > ACK of leg 2 and 200 OK of leg1 must be addressed to Opensips port 5061? > > Best Regards, > Sergey Pysanko. > > On Wed, Jan 5, 2022, 15:54 Bogdan-Andrei Iancu <bog...@opensips.org> > wrote: > >> Hi Sergey, >> >> If Asterisk is the one changing (from 5061 to 48470) the port in the >> RR/Route header, that's illegal to do. >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS eBootcamp 2021 >> https://opensips.org/training/OpenSIPS_eBootcamp_2021/ >> >> On 1/5/22 10:48 AM, Sergey Pisanko wrote: >> >> Hi, Bogdan. >> >> Yes, you are right. That's full call's scheme. >> >> Opensips:48470 Asterisk (5062) >> 1 leg ------------------INVITE (RR:5061)------------> >> <-----------------INVITE--------------------------------- 2 leg >> 2 leg --------------OK (RR:5061)--------------------> >> <--------------------ACK (Route:48470)------------ 2 leg >> < -------------------OK (RR: 48470) ----------------- 1 leg >> 1 leg. ACK From UA1 to Asterisk through Opensips (Route:48470) sent, but >> dropped. >> >> >> Best Regards, >> Sergey Pysanko. >> >> >> >> [image: Mailtrack] >> <https://mailtrack.io?utm_source=gmail&utm_medium=signature&utm_campaign=signaturevirality11&> >> Sender >> notified by >> Mailtrack >> <https://mailtrack.io?utm_source=gmail&utm_medium=signature&utm_campaign=signaturevirality11&> >> 01/05/22, >> 10:45:28 AM >> >> вт, 4 янв. 2022 г. в 20:44, Bogdan-Andrei Iancu <bog...@opensips.org>: >> >>> Sergey, >>> >>> I see OpenSIPS sents to Asterisk in INVITE: >>> >>> Record-Route: >>> <sip:Opensips_IP:5061;transport=tls;lr;ftag=d8e0d49a268d4b51aa85b8f79d2dc062> >>> >>> but in the 200 reply from Asterisk back to OpenSIPS I see: >>> >>> Record-Route: >>> <sip:Opensips_IP:48470;transport=TLS;lr;ftag=d8e0d49a268d4b51aa85b8f79d2dc062> >>> >>> Is asterisk the once changing the port there ??? >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> https://www.opensips-solutions.com >>> OpenSIPS eBootcamp 2021 >>> https://opensips.org/training/OpenSIPS_eBootcamp_2021/ >>> >>> On 1/4/22 3:11 PM, Sergey Pisanko wrote: >>> >>> Hi, Bogdan. >>> >>> Here is my simple scenario description: >>> >>> UA1----Opensips----Asterisk ---- Opensips ----UA2 >>> >>> Transport protocol doesn't change during this chain and it's tls, if I >>> understand you right. >>> >>> I attached SIP capture of the call. As you can see, there is the >>> dynamic tcp port in the RR hrd of last reply to client from which Opensips >>> connected to the Asterisk. Instead of one, to which UA1 connected to >>> Opensips (5061). As a result, there is a media session between UAs, but >>> only for 30 sec, during of which the UA1 tried to send ACK to the Opensips, >>> but unsuccessfully for quite clear reason. Is there the resolution how to >>> realize this scenario without rewriting RR? >>> >>> Best Regards, >>> Sergey Pysanko. >>> >>> >>> >>> >>> >>> >>> [image: Mailtrack] >>> <https://mailtrack.io?utm_source=gmail&utm_medium=signature&utm_campaign=signaturevirality11&> >>> Sender >>> notified by >>> Mailtrack >>> <https://mailtrack.io?utm_source=gmail&utm_medium=signature&utm_campaign=signaturevirality11&> >>> 01/04/22, >>> 01:46:49 PM >>> >>> вт, 4 янв. 2022 г. в 11:47, Bogdan-Andrei Iancu <bog...@opensips.org>: >>> >>>> Hi Sergey, >>>> >>>> Manually altering the RR hdr is a receipt for disaster :). Somehow I >>>> suspect you do not do double RR (as the protocol changes for the call). >>>> This double RR is automatically done (by default) when doing >>>> `record_route()`. Do you get 2 RR hdrs when routing the initial INVITE ? >>>> >>>> Regards, >>>> >>>> Bogdan-Andrei Iancu >>>> >>>> OpenSIPS Founder and Developer >>>> https://www.opensips-solutions.com >>>> OpenSIPS eBootcamp 2021 >>>> https://opensips.org/training/OpenSIPS_eBootcamp_2021/ >>>> >>>> On 1/4/22 11:27 AM, Sergey Pisanko wrote: >>>> >>>> Hello, Bogdan, . >>>> >>>> Thank you for your answer. I've solved my issue recently just rewriting >>>> Record - Route header with appropriate port within "onreply route block" by >>>> subst function. >>>> >>>> Best Regards, >>>> Sergey Pysanko. >>>> >>>> >>>> >>>> [image: Mailtrack] >>>> <https://mailtrack.io?utm_source=gmail&utm_medium=signature&utm_campaign=signaturevirality11&> >>>> Sender >>>> notified by >>>> Mailtrack >>>> <https://mailtrack.io?utm_source=gmail&utm_medium=signature&utm_campaign=signaturevirality11&> >>>> 01/04/22, >>>> 11:27:07 AM >>>> >>>> пн, 3 янв. 2022 г. в 17:59, Bogdan-Andrei Iancu <bog...@opensips.org>: >>>> >>>>> Hello Sergey, >>>>> >>>>> Could you provide a SIP capture (and calling scenario) to underline >>>>> the issue you have ? >>>>> >>>>> Best regards, >>>>> >>>>> Bogdan-Andrei Iancu >>>>> >>>>> OpenSIPS Founder and Developer >>>>> https://www.opensips-solutions.com >>>>> OpenSIPS eBootcamp 2021 >>>>> https://opensips.org/training/OpenSIPS_eBootcamp_2021/ >>>>> >>>>> On 12/30/21 2:50 PM, Sergey Pisanko wrote: >>>>> >>>>> Hello! >>>>> >>>>> I try to realize the next scenario with UAs, Opensips-2.4 and Asterisk. >>>>> UAs are registered onto Asterisk through Opensips and also - on >>>>> Opensips if the 200 OK is came back from Asterisk. >>>>> Calls between UAs are relayed to Asterisk by Opensips. >>>>> This scenario works fine with udp. But it needs to do with tls. And >>>>> here I have the problem. What happens. >>>>> Unlike udp, tcp cannot listen its port and create clients connection >>>>> at the same time. Opensips listens tls port for clients connection >>>>> whereas it creates dynamic tcp port to connect to Asterisk. As a >>>>> result, I see that port in Record-Route header in 200 OK addressed to >>>>> caller. >>>>> Thus, callers ACK comes to that dynamic port instead of Opensips >>>>> listened port and Opensips dropped it. >>>>> And question is how to force Opensips to put right port for caller? >>>>> >>>>> Regards, >>>>> Serhii Pysanko. >>>>> >>>>> >>>>> >>>>> [image: Mailtrack] >>>>> <https://mailtrack.io?utm_source=gmail&utm_medium=signature&utm_campaign=signaturevirality11&> >>>>> Sender >>>>> notified by >>>>> Mailtrack >>>>> <https://mailtrack.io?utm_source=gmail&utm_medium=signature&utm_campaign=signaturevirality11&> >>>>> 12/30/21, >>>>> 02:49:47 PM >>>>> >>>>> _______________________________________________ >>>>> Users mailing >>>>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> >>>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users@lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>> >>> _______________________________________________ >>> Users mailing >>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users@lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> >> _______________________________________________ >> Users mailing >> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> > _______________________________________________ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >
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