[Asterisk-Users] asterisk sample size adjustment

[trixter aka Bret McDanel]

Is there any way to adjust the sample size asterisk uses for VoIP
codecs?  From what I have gathered it uses a fixed 20ms sample size for
all codecs.  While some require at least this, some can be configured
for less.  This results in more overhead, but can be tweaked to provide
more efficient transfer on the backbone links due to ATM framing
properties.

If anyone has any information on how to change the sample size I would
appreciate hearing about it, because I cant find anything with google.
Asterisk is a particularly bad google term since it is used as a
footnote market, wildcard, etc :P


-- 
Trixter http://www.0xdecafbad.com     Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group
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