[opus] Transcoding OPUS?
butrus.butrus at gmail.com
Sat Jul 27 11:18:13 PDT 2013
I'd like to ask whether there is some documentation with recommended
parameters for transcoding voice codecs such as G722, G711a/u <-> Opus with
My Idea is to have something like:
HW-Phone <-> Asterisk <---------> Asterisk <-> HW-Phone
in order to lower the bandwidth between the two Asterisk* servers while not
compromising (much) the quality of the call.
(* I mean of course a custom build of Asterisk patched with Opus support
and tuned for this task)
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