[Freeswitch-users] PA and ALSA

[Frank Rosengart]

Hi,

I' trying to use FS as a softphone, because it seems to be the only SIP
UA, with up-to-date and full CELT support (thanks for that, great work!)

But I'm really struggling with my audio config. I'm using Ubuntu
10.10, x86 with Gnome -> Pulseaudio.

1. mod_alsa simply segfaults immediately after receiving a call
'application="bridge" data="alsa"' or  'alsa call 123' from the console

2. I am running FS on two different computers. One, which is upgraded
from earlier times, portaudio works ok (not perfect, but I think that
are Alsa related issues). Another computer, with a fresh 10.10 install,
mod_portaudio can not find any audio devices.
mod_portaudio.so links libasound, so I assume that Alsa is included.
aplay -L lists available devices:

pulse
    Playback/recording through the PulseAudio sound server
front:CARD=NVidia,DEV=0
    HDA NVidia, ALC889A Analog
    Front speakers
...

Any ideas what is missing here?

3. Has the portaudio lib, which included with FS, the PA 'dmix patch'
applied? http://www.portaudio.com/trac/ticket/31

4. Has anyone started to build mod_pulseaudio ?
I know, there are latency issues, but we are talking about VoIP with
codec with frame sizes of 10ms and more.. Portaudio is really not the
first choice for sound APIs on Linux anymore...

5. Same config, using CELT on Windows, I get noise glitches every second
when calling an FS server, answering with the echo app. They are very
regular, so it's maybe a sample rate issue somewhere in the path...


Thanks for any hints.


Frank