Don't use sipp. Since every scenario in SIPP is actually programmed by you, there's a chance you are not implementing the requests/replies properly. E.g.: incorrect or absent tags, incorrect Vias, Routes or Record-Routes, incorrect CSeq, etc. Use a B2BUA as you UAS. Use Asterisk or FreeSwitch to make sure the signaling is properly implemented. Sniff the traffic with Wireshark or ngrep if using a B2BUA is not a option. I'm sure there's something wrong with the protocol. On Thu, Jan 14, 2016 at 4:14 PM, Joel Dodson <jdodson at acm.org> wrote: > Hi Matthew, > > python/pjsua2 might not be the best choice for writing a SIP server. > Several years ago, on PJSIP 0.8 to 1.0, I wrote a signaling and media > gateway. The advice then was to use the lower level libraries (PJSIP and > PJMEDIA is what I recall using). There are still some nice abstractions > and state machines at that level, but you have more flexibility with the > number calls you need to support. Now I'm working with servers externally > so using the python bindings is working very well for me. > > I suspect, as Carlos mentioned, you're running into issues with the > protocol. If you try to cancel a call before being answered (sending > CANCEL because you haven't received a 200 OK yet for the INVITE), the call > object is probably intentionally kept around to deal with any 200 OKs that > might be coming in for the original INVITE. When you call hangup on the > call object it probably results in a CANCEL or a BYE depending on the state > of the dialog. Prior to confirmed, it should send a CANCEL. It's been a > long time since I've read any SIP RFCs, or been in the PJSIP internals, so > I could be off a bit. > > hope that helps > > Joel > > > On Thu, Jan 14, 2016 at 2:03 PM, Carlos Ruiz D?az < > carlos.ruizdiaz at gmail.com> wrote: > >> Looks like your problem isn't pjsip related but SIP related. If your 30 >> seconds are actually exactly 32 seconds, then I'm correct. >> >> Try to make your scenario work with CSIPSimple (based on pjsip) on >> Android. >> >> Setup on it your SIP credentials, place the call and see if it works. If >> it does, then you have a problem with your code, if it doesn't, then you >> have a problem with your SIP server. >> >> On Thu, Jan 14, 2016 at 3:57 PM, Matthew Williams <mgwilliams at gmail.com> >> wrote: >> >>> Is there anyone on this list with an in-depth understanding of the >>> entire stack that would be interested in doing some paid consulting? >>> >>> I have made considerable progress in getting an app running using the >>> swig/python bindings for pjsua2. However, due to the limited documentation, >>> it is slow going -- for example, it took quite some time to track down the >>> cause of segfaults to the Call object being lost from focus. >>> >>> My current issue is related to calls that are cancelled. It appears that >>> if hangup is called prior to the call entering the confirmed state, the >>> disconnected state is only entered some 30 seconds later. Since I am >>> writing a server application, this quickly eats up call limits. >>> >>> Please feel free to reply privately. >>> >>> Thanks! >>> >>> _______________________________________________ >>> Visit our blog: http://blog.pjsip.org >>> >>> pjsip mailing list >>> pjsip at lists.pjsip.org >>> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >>> >>> >> >> >> -- >> Carlos >> http://caruizdiaz.com >> >> _______________________________________________ >> Visit our blog: http://blog.pjsip.org >> >> pjsip mailing list >> pjsip at lists.pjsip.org >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > -- Carlos http://caruizdiaz.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20160114/54e61328/attachment.html>
