<div dir="ltr"><div><div><div><div><div><div><div>Hello,<br><br></div>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.<br>
<br></div>My Idea is to have something like:<br><br></div> HW-Phone <-> Asterisk <---------> Asterisk <-> HW-Phone<br></div> (G722) (Opus) (G722)<br>
<br></div>in order to lower the bandwidth between the two Asterisk* servers while not compromising (much) the quality of the call.<br><br></div>Thanks!<br><br></div>(* I mean of course a custom build of Asterisk patched with Opus support and tuned for this task)<br>
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