Hi! Ticket 467 is still closed - it should be reopened due to my bug reports (see below). Is it possible for users to open/change track tickets or is this only allowed for core developers? regards klaus Klaus Darilion wrote: > > Nanang Izzuddin schrieb: >> Hi Klaus, >> >> Ticket 467 is already fixed and in the trunk. >> If you have a chance, please do some tests and report back the result. > > Hi Nanang! > > I just made some test and now pjsua automatically hangs up the call, but: > > 1. The BYE is sent before the ACK (from INV-200). I think this is valid > but IMO it would be nicer if BYE is sent after ACK. > > 2. The BYE contains a session description, this is IMO a bug: > > BYE sip:klaus.darilion at 10.10.33.21:5060 SIP/2.0 > Via: SIP/2.0/UDP > 10.10.0.51:3891;rport;branch=z9hG4bKPj096a048772764b659103cb06bc9b80ff > Max-Forwards: 70 > From: sip:klaus.darilion@xxxxxxxxxxx;tag=2e731928dd334b91b716f476058249ac > To: sip:klaus.darilion at nic.at43.at;tag=000cce3a7bf800037b595478-6250939a > Call-ID: e20d9cd2582347d19032264cf02b5461 > CSeq: 5060 BYE > Route: <sip:10.18.53.113:6060;lr> > Route: <sip:10.10.32.160;lr;ftag=2e731928dd334b91b716f476058249ac> > Content-Type: application/sdp > Content-Length: 418 > > v=0 > o=- 3412948447 3412948447 IN IP4 10.10.0.51 > s=pjmedia > c=IN IP4 10.10.0.51 > t=0 0 > m=audio 4002 RTP/AVP 103 102 104 117 3 0 8 101 > a=rtcp:4003 IN IP4 10.10.0.51 > a=rtpmap:103 speex/16000 > a=rtpmap:102 speex/8000 > a=rtpmap:104 speex/32000 > a=rtpmap:117 iLBC/8000 > a=fmtp:117 mode=30 > a=rtpmap:3 GSM/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=sendrecv > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > > regards > klaus > > > > >> Thanks. >> >> cheers, >> nanang >> >> >> On 14/02/2008, *Nanang Izzuddin* <nanang.izzuddin at gmail.com >> <mailto:nanang.izzuddin at gmail.com>> wrote: >> >> Hi Klaus, >> >> No.1 is added to the existing ticket, I think the cases are related: >> http://trac.pjsip.org/repos/ticket/467 >> >> No.2 should be good and useful. >> https://trac.pjsip.org/repos/ticket/479 >> >> Thanks for the reports (& request :D) >> >> >> nanang >> >> >> >> On 14/02/2008, Klaus Darilion <klaus.mailinglists at pernau.at >> <mailto:klaus.mailinglists at pernau.at>> wrote: >> > Hi! >> > >> > I did some SRTP tests and it mostly works, but: >> > >> > 1. It does not work if pjsip sends SAVP and 200 OK contains AVP >> - then >> > pjsip has an open call but does not send RTP. The logs say: >> > >> > 16:02:54.437 pjsua_call.c SDP negotiation has failed: SDP media >> > transport type mismatch in offer/answer (PJMEDIA_SDPNEG_EINVANSTP) >> > [status=220046] >> > >> > What should happen now? Should pjsua-lib hang up the call? It >> does not >> > and the call changes to "confirmed". >> > >> > If the application should send the BYE, then there should be some >> > indication, e.g. PJSUA_CALL_MEDIA_SAVP_FAILED in the >> > call_media_state callback. Or is there already a callback which >> tell the >> > application about the SRTP failure? >> > >> > 2. If SRTP is optional and a call is established, it would be >> good to >> > retrieve SRTP status from pjsua (e.g. pjsua_call_get_info()) to >> indicate >> > SRTP status to the user. >> > >> > thanks >> > Klaus >> > >> > >> > _______________________________________________ >> > 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
