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. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org
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