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
