Hi
I have reading the voiip side i found some document says
"
The conference bridge runs Ulaw codec by default. If you let people connect
with GSM or other codecs, Asterisk will use CPU power to convert audio
between codecs "
iam using vicidial and meetme for callcenter application. iam geting choppy
voice, and voice breaks.
iam using connecting VoIP SIP provider using g729 codec, since i can save
bandwidth
iam using client side also g729, so no translation required
but after i see this document, will meetme convert the g729 to GSM or ULAW
internall, and
i have will have cpu load, is this correct.
if i dont want to CPU loadup more, i should use GSM or ULAW at client side
is this correct.
can some one correct me if iam wrong
suggestions welcome
ram
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