Hi, The sndtest report for device 'default' shows problem in the duration: Duration: 1s.000 while normally the duration should be around 10s. Please run sndtest for other device id (e.g: device id 12: /dev/dsp) and as pjsua by default running on 16kHz, I think it will better to run it on 16kHz as well, please check 'sndtest --help' for available params. Once you get a working device id, you should change your pjsua config, e.g: --capture-dev=12 --playback-dev=12 Regards, nanang On Fri, Oct 31, 2008 at 11:50 PM, emanuele bottegoni <emanuele.b at automaonline.com> wrote: > 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 > > > > _______________________________________________ > 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 >
