On Nov 26, 2003, at 10:34 AM, Peter Zeltins wrote:


I'm interested. I'm running chan_capi 0.3.0 with Fritz PCI ISDN card.
Using
DIAX as softphone and dialing out to PSTN generally results in good
sound
quality at softphone end (no echo), but PSTN end experiences quite a bit
of
echo. I have enabled echosquelch in capi.conf, but it does not seem to
help

Then my idea is not your solution, nor is echosquelch, I guess. Yes, it
would be possible to kludge chan_capi to alleviate your problem; no, I
do not think it would be the right solution. If the far end hears echo,
it's coming from the near end, that is from your softphone. You could
try to use something else to confirm that (a hardphone?). I suggest
that you play with the mixer settings on your own machine, because it's
your sound card that's recording the far end's voice!

I do not have a hardphone to play around with, but the echo is there both
with built-in audio card (SigmaTel) and Bluetooth headset. There are no
mixer settings than I can adjust as well. I'll try disabling AGC and/or
lowering mike sensitivity.

DIAX is built on my iaxclient library, and the library itself should not produce echo, although echo can be produced either by misconfigured Windows sound mixer settings (in some cases, the drivers don't make it possible to configure them to _not_ mix outbound audio back in with inbound), or acoustic echo.


The library has several DSP features, including AGC, denoising, and echo cancellation. These are all provided via integration with preprocessing from the SPEEX library. I don't know if DAN allows you to turn on/off echo cancellation or not. However, the echo cancellation code from speex is still very immature, and not quite there yet. In particular, it is almost completely ineffective is AGC is also being used, because the echo canceller leaves a smaller residual echo in the signal, and the AGC then boosts up the residual echo to full volume.

I've also tried integrating the various echo cancellers from asterisk into the code, but they are even less effective when used with the long echo tails seen in a VoIP client.

One thing you can do if you can't configure your card properly is use a "push to talk" feature. This would need to be implemented in DIAX, although example code exists in other clients. Basically, this looks for the user to press some "hotkey" before speaking (like a walkie-talkie"), and keeps audio muted unless the key is pressed.

-SteveK


_______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to