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hi,<br>
<br>
723 needs 30ms + 5ms forward. 729 needs 10ms + 5ms forward. The
'standard' for 711 is actually 6ms (48 bytes). This would have to be
done per channel (or per codec), but I am not sure wherever Asterisk
allow per codec size or run's with one static size???<br>
<br>
Jan<br>
<br>
trixter aka Bret McDanel wrote:
<blockquote cite="mid1131980304.8892.24.camel@rufus.home.tld"
type="cite">
<pre wrap="">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
</pre>
<pre wrap="">
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