[opus] Transcoding OPUS?

[Butrus Damaskus]

Hello,

I'd like to ask whether there is some documentation with recommended
parameters for transcoding voice codecs such as G722, G711a/u <-> Opus with
near-transparency.

My Idea is to have something like:

 HW-Phone  <-> Asterisk   <--------->  Asterisk <-> HW-Phone
  (G722)
(Opus)                             (G722)

in order to lower the bandwidth between the two Asterisk* servers while not
compromising (much) the quality of the call.

Thanks!

(* I mean of course a custom build of Asterisk patched with Opus support
and tuned for this task)
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