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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