Consulting Needed

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Hi Walter,

That was exactly it - I was not replying with the 487 after receiving the
CANCEL. Thanks!

Matthew

On Fri, Jan 15, 2016 at 4:48 PM, Walter Doekes <walter+pjsip at wjd.nu> wrote:

> Matthew,
>
> do they really stay active after a complete cancel? Or is that just
> because you're using the wrong SIPp scenario and the pjsip breaks down
> the call first after lack of activity after the 32 seconds?
>
> I couldn't tell from your responses whether your server scenario was
> properly adjusted to handling the cancel.
>
> If not, try this scenario:
>
> https://github.com/ossobv/sipp-scenarios/blob/master/out-in/iinv-o180-icancel.xml
>
> It expects an INVITE (i), sends 180 (o) and expects a CANCEL (i); to
> which it properly replies with a 200, and a 487 to the INVITE.
>
> (The scenario's may require a recent (master) version of sipp.)
>
> Cheers,
> Walter Doekes
> OSSO B.V.
>
>
> On 14-01-16 23:47, Matthew Williams wrote:
> > pjsip has a max_calls configuration (and a corresponding constant which
> > sets a hard limit in the c headers). I had this set to 512, which was
> > being hit very quickly since in this particular scenario calls are
> > staying "active" for 32s when I was expecting them to only remain active
> > for < 1s due to the immediate CANCEL. It was really all caused by a
> > misunderstanding of the RFC and not with the application or signalling
> > per se.
> >
> > On Thu, Jan 14, 2016 at 5:36 PM, Carlos Ruiz D?az
> > <carlos.ruizdiaz at gmail.com <mailto:carlos.ruizdiaz at gmail.com>> wrote:
> >
> >     Not sure if I understand what you mean with increasing the active
> >     calls limit.
> >
> >     The solution is these cases is to figure out what went wrong with
> >     the signaling, and fixing it in the SIP level.
> >
> >     Looks like you have identified the problem and the potential
> >     solution. Make sure to update the thread with your results :).
> >
> >     Regards,
> >     Carlos
> >
> >     On Thu, Jan 14, 2016 at 4:28 PM, Matthew Williams
> >     <mgwilliams at gmail.com <mailto:mgwilliams at gmail.com>> wrote:
> >
> >         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 <mailto: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 <mailto: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
> >                 <mailto: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 <mailto: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 <mailto:
> 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 <mailto: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 <mailto: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 <mailto: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 <mailto: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 <mailto: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
> >
>
>
> _______________________________________________
> 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
>
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