[Asterisk-Dev] Help Debugging Dropped Call Audio

[Steven Critchfield]

On Tue, 2005-12-27 at 11:40 -0800, Mike Benoit wrote:
> I got my hands on a couple of the raw .wav files, and it seems they do
> not contain the artifacts I described in earlier emails. I ran my mp3
> conversion script on the .wav's and the resulting mp3's do have the
> artifacts (if played in XMMS only). So unless the MP3 conversion is
> somehow picking up dropped audio packets which XMMS is exposing, this
> doesn't appear to be the same issue you are running in to.

First off, mp3 is a BAD choice for telephony audio storage. It doesn't
compress very much compared to other options available to you and the
sound quality doesn't make enough of a trade off for the size of the
file. 

Second, are you experiencing the audio problems on raw wav files? It
very well may be that the interaction between your encoder and xmms is
what is causing your problems. 
-- 
Steven Critchfield <critch at basesys.com>