[SIP SIMPLE client] Audio problems

[Santiago Gimeno]

Hello,

I've been using the SDK to develop an application for a couple of months and
so far so good until audio problems issues have appeared.

When a call is established between the application and another softphone,
sometimes an annoying buzz is heard on the softphone end.
I have captured the rtp traffic and the noise is introduced in the side of
the application.
Then, I wanted to check if the audio was produced by the sound card or the
application. I'm using a handset device with a sound card integrated as
input and output device, and I couldn't play or record any data at the same
time the device was being used by the application. So I have followed the
tutorial on: http://sipsimpleclient.com/wiki/SipAudioDeviceSharingLinux, and
another issue have appeared: while I can play lots of files simultaneously
using aplay: for example: "$ aplay -D default
/usr/share/blink/sounds/beeping_ringtone.wav", the application doesn't play
audio anymore, I even though I have configured both input_device and
output_device to use the default device.
By checking the pjsip trace I can't see anything wrong, it looks like the
device is opened with no errors:


2011-01-19 13:22:50.853000 [phone.pyc 2187] (4)      pasound.c: PortAudio
sound library initialized, status=0
2011-01-19 13:22:50.853000 [phone.pyc 2187] (4)      pasound.c: PortAudio
host api count=2
2011-01-19 13:22:50.853000 [phone.pyc 2187] (4)      pasound.c: Sound device
count=15
2011-01-19 13:22:50.880000 [phone.pyc 2187] (5)      pasound.c: Opened
device default(ALSA)/default(ALSA) for recording and playback, sample
rate=44100, ch=1, bits=16, 882 samples per frame, input latency=100 ms,
output latency=100 ms
2011-01-19 13:22:50.880000 [phone.pyc 2187] (5)      pasound.c: Starting
default stream..
2011-01-19 13:22:50.880000 [phone.pyc 2187] (5)      pasound.c: Done,
status=0
2011-01-19 13:22:50.890000 [phone.pyc 2187] (5)      pasound.c: Opened
device 14: default(ALSA) for playing, sample rate=44100, ch=1, bits=16, 882
samples per frame, latency=100 ms
2011-01-19 13:22:50.890000 [phone.pyc 2187] (5)      pasound.c: Starting
default stream..
2011-01-19 13:22:50.890000 [phone.pyc 2187] (5)      pasound.c: Done,
status=0
2011-01-19 13:22:50.945000 [phone.pyc 2187] (5)      pasound.c: Player
thread started
2011-01-19 13:22:50.981000 [phone.pyc 2187] (5)      pasound.c: Recorder
thread started
2011-01-19 13:22:50.981000 [phone.pyc 2187] (5)      pasound.c: Player
thread started


And when a try to play a wav file, no errors are reported:


==> /home/sgimeno/.phone/notifications_trace.txt <==
2011-01-19 13:30:31.989499 [phone.pyc 2187]: Notification
name=AudioPortDidChangeSlots sender=<sipsimple.audio.WavePlayer object at
0xa74ad2c>
{'consumer_slot_changed': False,
 'new_producer_slot': None,
 'old_producer_slot': None,
 'producer_slot_changed': True,
 'timestamp': datetime.datetime(2011, 1, 19, 13, 30, 31, 988983)}
2011-01-19 13:30:31.989700 [phone.pyc 2187]: Notification
name=WavePlayerDidEnd sender=<sipsimple.audio.WavePlayer object at
0xa74ad2c>
{'timestamp': datetime.datetime(2011, 1, 19, 13, 30, 31, 989090)}
2011-01-19 13:30:31.990557 [phone.pyc 2187]: Notification
name=WavePlayerDidStart sender=<sipsimple.audio.WavePlayer object at
0xa1d5fec>
{'timestamp': datetime.datetime(2011, 1, 19, 13, 30, 31, 990257)}
2011-01-19 13:30:31.991000 [phone.pyc 2187]: Notification
name=AudioPortDidChangeSlots sender=<sipsimple.audio.WavePlayer object at
0xa1d5fec>
{'consumer_slot_changed': False,
 'new_producer_slot': 1,
 'old_producer_slot': None,
 'producer_slot_changed': True,
 'timestamp': datetime.datetime(2011, 1, 19, 13, 30, 31, 990306)}
2011-01-19 13:30:31.991224 [ionphone.pyc 2187]: Notification name=Play
sender=<InputThread(Thread-2, started daemon -1231606928)>
{'count': -1, 'pause': -1, 'tone': 'dtmf_1_tone', 'volume': 70}
2011-01-19 13:30:32.042914 [ionphone.pyc 2187]: Notification
name=WaveFileDidFinishPlaying sender=<sipsimple.core._core.WaveFile object
at 0xa70c42c>
{'timestamp': datetime.datetime(2011, 1, 19, 13, 30, 31, 988961)}

==> /home/sgimeno/.phone/pjsip_trace.txt <==
2011-01-19 13:30:31.990000 [phone.pyc 2187] (4)   wav_player.c: File player
'/usr/share/blink/sounds/dtmf_1_tone.wav' created: samp.rate=32000, ch=1,
bufsize=4KB, filesize=32KB
2011-01-19 13:30:31.990000 [phone.pyc 2187] (5)     resample.c: resample
created: high qualiy, large filter, in/out rate=32000/44100
2011-01-19 13:30:31.990000 [phone.pyc 2187] (5)     resample.c: resample
created: high qualiy, large filter, in/out rate=44100/32000
2011-01-19 13:30:31.990000 [iphone.pyc 2187] (4)   conference.c: Port 1
(/usr/share/blink/sounds/dtmf_1_tone.wav) transmitting to port 0
(Master/sound)


In sort, I'm a bit lost and any ideas on these issues would be much
appreciated.

Thanks,

Best regards,

Santi
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