Okay. But lets say i would not be on the RPi but just on linux and i would have the same problem with the “No default audio device” any idea where this could come from? Have you checked my last logs? Is the codec thing correct now? > Am 04.08.2017 um 07:20 schrieb Andreas Wehrmann <a.wehrmann@xxxxxxxxxx>: > > On 08/04/2017 07:03 AM, Kevin Rombach via pjsip wrote: >> For me it looks like there is only the 1 codec now in the SDP message. But you can look in the logs below and tell me if im right? ;) >> >> Have you any suggestions for me for my other problem with the sound: >> >> Unable to find default audio device <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2017-August/020128.html> >> >> Sideinfo: Im using the pjlib cross compiled for an raspberry pi 3 and i am using it with Qt. >> > Sorry no clue, I haven't played around with RPi and PJ yet. > >> 06:59:59.079 pjsua_core.c .RX 707 bytes Response msg 183/INVITE/cseq=10061 (rdata0x7550169c) from UDP 192.168.178.1:5060: >> SIP/2.0 183 Session Progress >> Via: SIP/2.0/UDP 192.168.178.42:5060;rport=5060;branch=z9hG4bKPjzuUpsmNa3R2odjt52f-Zn0ogRbiQ0auT >> From: <sip:doorz-control@xxxxxxxxx>;tag=JtPN-VUUOmhq3.I4c4KkZcjZu-xz.95D >> To: <sip:**1@xxxxxxxxx>;tag=CC3E030550BCB9D5 >> Call-ID: jH1tw2KBSeczVda3MoNbKcsViQAFvzVL >> CSeq: 10061 INVITE >> Contact: <sip:EEE303552C7E89C15FFEDA99CA2A7@192.168.178.1> >> User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.06.83 (Mar 8 2017) >> Content-Type: application/sdp >> Content-Length: 216 >> >> v=0 >> o=user 4919567 4919567 IN IP4 192.168.178.1 >> s=pjmedia >> c=IN IP4 192.168.178.1 >> t=0 0 >> m=audio 7078 RTP/AVP 9 96 >> a=rtpmap:9 G722/8000 >> a=rtpmap:96 telephone-event/8000 >> a=fmtp:96 0-15 >> a=sendrecv >> a=rtcp:7079 >> >> --end msg-- >> 06:59:59.079 pjsua_media.c .....Call 0: updating media.. >> 06:59:59.079 pjsua_aud.c ......Audio channel update.. >> 06:59:59.080 strm0x7550836c .......VAD temporarily disabled >> 06:59:59.080 strm0x7550836c .......Encoder stream started >> 06:59:59.080 strm0x7550836c .......Decoder stream started >> 06:59:59.080 pjsua_media.c ......Audio updated, stream #0: G722 (sendrecv) >> 06:59:59.080 pjsua_aud.c .....Conf connect: 1 --> 0 >> 06:59:59.080 conference.c ......Port 1 (sip:**1@xxxxxxxxx) transmitting to port 0 (Master/sound) >> 06:59:59.491 stream.c G722 codec used, remote samples per frame detected = 80 >> 06:59:59.718 strm0x7550836c VAD re-enabled >> 07:00:08.602 pjsua_core.c .RX 803 bytes Response msg 488/INVITE/cseq=10061 (rdata0x7550169c) from UDP 192.168.178.1:5060: >> SIP/2.0 488 Not Acceptable Here >> Via: SIP/2.0/UDP 192.168.178.42:5060;rport=5060;branch=z9hG4bKPjzuUpsmNa3R2odjt52f-Zn0ogRbiQ0auT >> From: <sip:doorz-control@xxxxxxxxx>;tag=JtPN-VUUOmhq3.I4c4KkZcjZu-xz.95D >> To: <sip:**1@xxxxxxxxx>;tag=CC3E030550BCB9D5 >> Call-ID: jH1tw2KBSeczVda3MoNbKcsViQAFvzVL >> CSeq: 10061 INVITE >> Warning: 399 0.0.0.0 "successful but result empty" >> User-Agent: FRITZ!OS >> Content-Type: application/sdp >> Content-Length: 359 >> >> v=0 >> o=user 4919567 4919568 IN IP4 192.168.178.1 >> s=call >> c=IN IP4 192.168.178.1 >> t=0 0 >> m=audioMyCall::onCallState >> > > Hmm, Fritzbox now rejects the call... this is weird. > I'd say you go ahead and disable all but PCMA/8000 and see if THAT works; just to make sure it works at all... > > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@xxxxxxxxxxxxxxx > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org _______________________________________________ Visit our blog: http://blog.pjsip.org pjsip mailing list pjsip@xxxxxxxxxxxxxxx http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
