Hi Joshua, Am Montag, dem 21.03.2022 um 07:34 -0300 schrieb Joshua C. Colp: > On Mon, Mar 21, 2022 at 7:18 AM Karsten Wemheuer <k...@mail.de> > wrote: > > Hi *, > > > > i am trying to analyze a problem with pjsip. > > > > Scenario: Phones are registered to opensips. From there the calls > > go to > > asterisk and then on via the trunk. This works fine. > > > > In the opposite direction there is sometimes a problem: > > A call comes in over the trunk, asterisk sends the INVITE to > > opensips. > > From there the INVITE goes to the phone. After the call is answered > > (200 OK from phone via proxy), asterisk sends the ACK not via the > > proxy > > but directly to the phone. Looking at the debug log it looks like > > the > > destination address of the ACK is obtained from the Contact or RTP > > data > > and not from the Via header. > > > > I would like to check the source code to see if I am doing > > something > > wrong or if there is a bug. Where do I enter to investigate the > > construction of the ACK packet? > > I would not suggest looking at the source code for this. You would > still have to understand SIP itself to know what is going on, so the > RFC is really the best place. > > For your specific issue - unless the proxy is doing record routing, > then the behavior is correct. The Contact header would be used for > sending the ACK. RTP is never used for SIP signaling destination. >
Thanks a lot! The hint with record routing was very helpful. I have it working now. I just was confused by the debug log which lead me in the wrong direction. Have a nice day. Best regards Karsten -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
