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 <[email protected]>: > 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 <[email protected]>: > >> 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 <[email protected]>: >> >>> 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 >>> [email protected]http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >> _______________________________________________ >> Users mailing list >> [email protected] >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > _______________________________________________ > Users mailing > [email protected]http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >
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