Carlos, If I'm reading http://tools.ietf.org/html/rfc3261#section-17.1.1.2 correctly, then sipp & pjsip are both behaving as expected, and the solution is to just have a higher limit on active calls. I will certainly be testing with some other UAs, but sipp is convenient for load testing & developing against random scenarios. On Thu, Jan 14, 2016 at 5:23 PM, Carlos Ruiz D?az <carlos.ruizdiaz at gmail.com > wrote: > 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 > > _______________________________________________ > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20160114/26ad266b/attachment.html>
