Hi all.
I am trying to implement a phone system, using one Sheeva with asterisk as server, and one Sheeva with linphone as SIP client. To the client I have connectet a Native Union handset
http://www.nativeunion.com/us/wiredproducts/mmoshi-02.html. The conecction to the handset is a Native Union USB dongle sound card
http://www.nativeunion.com/us/accessories/usb-adp/usb-adaptor.html.
Using lsusb, the sound card reports as:
$this->bbcode_second_pass_code('', 'Bus 001 Device 008: ID 0c76:1607 JMTek, LLC. audio controller').
This device has hw support for 48000 Hz sample rate
When playing 48000 Hz streams (using mpg123) all is ok. playing 44100 streams is also ok (with a cpu load of approx 4%).
When receiving calls (not absolutetly certain about sample rate, but my guess is 8000 Hz) I get buffer overruns in ALSA, and the sound is delayed for playback, but not for capture) Capture works for some seconds, before everything goes heywire...
My /etc/asound.conf:
$this->bbcode_second_pass_code('', 'pcm.moshi {
type hw
card Audio
}
pcm.linphoneout {
type plug
slave {
pcm "moshi"
rate 48000
}
}
pcm.linphonein {
type dsnoop
slave {
pcm "moshi"
rate 48000
}
bindings {
0 0
}
}
pcm.linphone {
type asym
playback.pcm "linphonein"
capture.pcm "linphoneout"
}
ctl.moshi {
type hw
card Audio
}
ctl.linphonein {
type hw
card Audio
}
ctl.linphoneout {
type hw
card Audio
}
')
I have a second usb sound card that is default. The config for that is not included. It makes no difference if is included or not.
I have tried some different codecs, but none works. I have tried linpone on a windows 7 machine connected to the same asterisk server, performes flawless.
Last note, when I answer a incomming call, the CPU load for linphone peaks to 99%
Could this be softfloat related (ALSA/linphone bitrate conversion )?
I am not completely lost, but realy running out of options.
Any toughts are very wellcome!