Notices

i am playing with default-fragments and default-fragment-size-msec , and i am getting real nice results ... still testing ....
changed default-fragments from 2 to 5.

I just found out that the sound is much less choppy using sip if the phone is overclocked to 900MHz. So it seems to be definitely a performance issue.
I have this choppy sound for WLAN and 3G sip calls. I use the g729 codec.

is it possible that the phone is doing transcoding when using g729 and can't keep up? i use g711 ulaw and have never experienced choppy sip audio and i use n900 everyday as office extension.

It seems that it is a codec (performance) issue. If I use the ulaw codec the audio is fine. However if I use g729 or ilbc I get choppy audio.
Because I also want to use sip via 3G/EDGE I have to use a codec that uses low bandwidth.

how did you do this?

to work around an issue concerning power saving and choppy audio which may be caused by a faulty AP or maybe by Constant Active Mode (CAM) I have good results by disabling power saving in the gui and managing power saving on wlan0 through iwconfig and dbus events.

this way I have a good battery lifetime when I put wlan0 in power saving mode if I do not use sip and I have no more latency problems if I disable power saving mode on wlan0 when I receive or initiate a sip call.

now i can make a phone call for about 15 min, after that the audio is choppy and I can't understand my opponent as described in the bug.

ah - ok! I found the place where you did this. you modified pulseaudio's daemon.conf.

Yep. Sorry the delay....

A little bit off topic but I don't want to open another thread for this. I have changed the codec once more and now use the speex codec. This seems to work without choppy audio.

Now the question: How can I change the settings of the speex codec? (Speex can be used for different bandwidth usages. There are quality and complexity parameter. Also there is a parameter to configure whether to use static or variable bitrate) Is there a config file on the n900 for this?
On the server side (asterisk) I can change the speex codecs settings in the codecs.conf settings file. But this is only for the audio stream from the server to the client.

i was thinking i might be the only one - i have choppy audio on sip calls as well. i didn't try skype. but sip definitely is choppy on my pr1.2 device - even with 850mhz and usually from the very beginning of the call no matter weather beeing used over wifi or 3g.
thank you.

I have to correct myself. The audio is also choppy after some time using speex. So the only codec that currently works perfect without choppy audio for me is ulaw (g711). However this codec uses too much bandwidth to work nicely on a EDGE network. :-(
All other codecs seems to consume too much cpu power or there is something wrong configured on a stock n900.