[pjsip] AUDIO Problem
Nanang Izzuddin
nanang at pjsip.org
Tue Nov 4 10:28:36 EST 2008
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
>
>
>
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