On 08/04/2017 07:24 AM, Kevin Rombach via pjsip wrote:
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?
Yes I did, see below; PJ correctly offers G.722 only.
I was wondering why the Fritzbox is now rejecting your call and
suggested you enable only PCMA/8000 to see if it works at all.
I checked your logs again and found that only the "NULL sound device" is
connected to your confbridge.
So it looks like there is no "real" sound port connected to any
confbridge port,
which would explain the lack of audio:
06:59:57.997 pjsua_aud.c Set sound device: capture=-99, playback=-2
06:59:57.997 pjsua_aud.c Set sound device: capture=-99, playback=-99
06:59:57.997 pjsua_aud.c .Setting null sound device..
06:59:57.997 pjsua_aud.c ..Opening null sound device..
Are you telling PJSUA to use the NULL device when setting it up?
See:
http://www.pjsip.org/docs/latest-2/pjsip/docs/html/group__PJSUA__LIB__MEDIA.htm#ga2e6cb631c6ca40d30973cc5ebeaba255
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...
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