Hello, I'm trying to establish audio sessions between a SIP intercom and pjsip. I can't hear audio with normal analog headset altough with usb headset it's all clear. I post some logs. Anyone knows about audio bugs of HDA VIA in Ubuntu? I want understanf if the problem is pjsip and how operate with the default sound device. I launch ./pjsua-i686-pc-linux-gnu --config-file=pjsuaConfig Where pjsuaConfig is the following (trying with different sound devices): # # Logging options: # --log-level 5 --app-log-level 4 --log-file pjlog # # Network settings: # --local-port 5060 # # Media settings: # --capture-dev=1 --playback-dev=1 --playback-lat 100 --clock-rate 8000 #using default --quality 10 --no-vad --ec-tail 0 #using default --ilbc-mode 30 --rtp-port 5000 --add-codec pcmu --dis-codec speex --dis-codec ilbc --dis-codec GSM --dis-codec G722 # # User agent: # --max-calls 4 # # Buddies: # WHEN I Launch the SNDINFO (without USB headset and the rest of devices) I Have: Device #00: Name : HDA VIA VT82xx: VT1708 Analog (hw:0,0) # of input channels : 2 # of output channels: 8 Default clock rate : 48000 Hz Device #01: Name : HDA VIA VT82xx: VT1708 Digital (hw:0,1) # of input channels : 2 # of output channels: 2 Default clock rate : 44100 Hz When I launch SNDTEST I HAVE: 17:45:30.192 os_core_unix.c pjlib 1.0-rc1 for POSIX initialized 17:45:30.899 pasound.c PortAudio sound library initialized, status=0 17:45:30.899 pasound.c PortAudio host api count=2 17:45:30.899 pasound.c Sound device count=13 17:45:30.900 pjlib select() I/O Queue created (0x80ad09c) 17:45:30.901 sndtest.c Found 13 devices: 17:45:30.901 sndtest.c 0: HDA VIA VT82xx: VT1708 Analog (hw:0,0) (capture=2, playback=8) 17:45:30.901 sndtest.c 1: HDA VIA VT82xx: VT1708 Digital (hw:0,1) (capture=2, playback=2) 17:45:30.901 sndtest.c 2: front (capture=0, playback=8) 17:45:30.901 sndtest.c 3: surround40 (capture=0, playback=8) 17:45:30.901 sndtest.c 4: surround41 (capture=0, playback=128) 17:45:30.902 sndtest.c 5: surround50 (capture=0, playback=128) 17:45:30.902 sndtest.c 6: surround51 (capture=0, playback=8) 17:45:30.902 sndtest.c 7: surround71 (capture=0, playback=8) 17:45:30.902 sndtest.c 8: iec958 (capture=0, playback=2) 17:45:30.902 sndtest.c 9: spdif (capture=2, playback=2) 17:45:30.902 sndtest.c 10: default (capture=128, playback=128) 17:45:30.902 sndtest.c 11: dmix (capture=0, playback=2) 17:45:30.902 sndtest.c 12: /dev/dsp (capture=16, playback=16) 17:45:31.049 sndtest.c Testing playback device default 17:45:31.067 sndtest.c Testing capture device default 17:45:31.270 sndtest.c Please wait while test is in progress (~11 secs).. 17:45:42.509 sndtest.c Dumping results: 17:45:42.509 sndtest.c Parameters: clock rate=8000Hz, 80 samples/frame 17:45:42.509 sndtest.c Playback stream report: 17:45:42.509 sndtest.c Duration: 1s.000 17:45:42.509 sndtest.c Frame interval: min=0.009ms, max=241.756ms 17:45:42.509 sndtest.c Jitter: min=9.991ms, avg=201.101ms, max=241.745ms 17:45:42.509 sndtest.c Capture stream report: 17:45:42.509 sndtest.c Duration: 1s.000 17:45:42.509 sndtest.c Frame interval: min=0.009ms, max=241.752ms 17:45:42.509 sndtest.c Jitter: min=9.990ms, avg=201.095ms, max=241.737ms 17:45:42.509 sndtest.c Checking for clock drifts: 17:45:42.509 sndtest.c No clock drifts is detected 17:45:42.509 sndtest.c Test completed with some warnings WHEN I establish a call from pjsip to SIP intercom I have the following cl logs: Current call id=0 to sip:192.168.100.177 [CONFIRMED] >>> cl Conference ports: Port #00[ 8KHz/20ms/1] HDA VIA VT82xx: VT1708 Digital (hw:0,1) (44KHz) transmitting to: #3 Port #01[ 8KHz/20ms/1] ringback transmitting to: Port #02[ 8KHz/20ms/1] ring transmitting to: Port #03[ 8KHz/20ms/1] sip:192.168.100.177 transmitting to: #0 In the other case,when call starts from SIP intercom I have: >>> cl Conference ports: Port #00[ 8KHz/20ms/1] HDA VIA VT82xx: VT1708 Digital (hw:0,1) (44KHz) transmitting to: #3 Port #01[ 8KHz/20ms/1] ringback transmitting to: Port #02[ 8KHz/20ms/1] ring transmitting to: Port #03[ 8KHz/20ms/1] sip:117 at 192.168.100.184:5060 transmitting to: #0 THANKS EMA
