[asterisk-users] opus codec

[Jerry Geis]

Hi All,

I am trying to get the opus codec working with linphone.
I followed the instructions... This shows me its loaded

core show translation paths opus
--- Translation paths SRC Codec "opus" sample rate 48000 ---
        opus:48000       To g723:8000       : No Translation Path
        opus:48000       To ulaw:8000       : (opus at 48000)->(slin at 48000
)->(slin at 8000)->(ulaw at 8000)
        opus:48000       To alaw:8000       : (opus at 48000)->(slin at 48000
)->(slin at 8000)->(alaw at 8000)
        opus:48000       To gsm:8000        : (opus at 48000)->(slin at 48000
)->(slin at 8000)->(gsm at 8000)
        opus:48000       To g726:8000       : (opus at 48000)->(slin at 48000
)->(slin at 8000)->(g726 at 8000)
        opus:48000       To g726aal2:8000   : (opus at 48000)->(slin at 48000
)->(slin at 8000)->(g726aal2 at 8000)
        opus:48000       To adpcm:8000      : (opus at 48000)->(slin at 48000
)->(slin at 8000)->(adpcm at 8000)
        opus:48000       To slin:8000       : (opus at 48000)->(slin at 48000
)->(slin at 8000)
        opus:48000       To slin:12000      : (opus at 48000)->(slin at 48000
)->(slin at 12000)
        opus:48000       To slin:16000      : (opus at 48000)->(slin at 48000
)->(slin at 16000)
        opus:48000       To slin:24000      : (opus at 48000)->(slin at 48000
)->(slin at 24000)
        opus:48000       To slin:32000      : (opus at 48000)->(slin at 48000
)->(slin at 32000)
        opus:48000       To slin:44100      : (opus at 48000)->(slin at 48000
)->(slin at 44100)
        opus:48000       To slin:48000      : (opus at 48000)->(slin at 48000)
        opus:48000       To slin:96000      : (opus at 48000)->(slin at 48000
)->(slin at 96000)
        opus:48000       To slin:192000     : (opus at 48000)->(slin at 48000
)->(slin at 192000)
        opus:48000       To lpc10:8000      : (opus at 48000)->(slin at 48000
)->(slin at 8000)->(lpc10 at 8000)
        opus:48000       To g729:8000       : No Translation Path
        opus:48000       To speex:8000      : (opus at 48000)->(slin at 48000
)->(slin at 8000)->(speex at 8000)
        opus:48000       To speex:16000     : (opus at 48000)->(slin at 48000
)->(slin at 16000)->(speex at 16000)
        opus:48000       To speex:32000     : (opus at 48000)->(slin at 48000
)->(slin at 32000)->(speex at 32000)
        opus:48000       To ilbc:8000       : (opus at 48000)->(slin at 48000
)->(slin at 8000)->(ilbc at 8000)
        opus:48000       To g722:16000      : (opus at 48000)->(slin at 48000
)->(slin at 16000)->(g722 at 16000)
        opus:48000       To siren7:16000    : No Translation Path
        opus:48000       To siren14:32000   : No Translation Path
        opus:48000       To testlaw:8000    : (opus at 48000)->(slin at 48000
)->(slin at 8000)->(testlaw at 8000)
        opus:48000       To g719:48000      : No Translation Path
        opus:48000       To none:8000       : No Translation Path
        opus:48000       To silk:8000       : No Translation Path
        opus:48000       To silk:12000      : No Translation Path
        opus:48000       To silk:16000      : No Translation Path
        opus:48000       To silk:24000      : No Translation Path

I set linphone to ONLY use opus codec. When I call in the call works - but
no audio.
Whats next?

Jerry
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20190708/531e8db0/attachment.html>