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From: | Anatoli |
Subject: | Re: [Linphone-users] Linphone opus and asterisk 13 |
Date: | Tue, 9 Jul 2019 07:05:07 -0300 |
Hi Jerry,
The translation paths you show are only for 48KHz opus, but your endpoints may use other rates. In any case you should enable logs and see what's going on: pjsip set logger on. Also, if both endpoints can use same rate opus, you may connect them without transcoding. Regards, Anatoli From:
Jerry Geis <address@hidden>
Sent: Monday, July 08, 2019 09:26 To: Linphone-users <address@hidden> Subject: [Linphone-users] Linphone opus and asterisk 13 I am trying to use linphone on windows with
Asterisk 13. I set my extension to use only opus, I set linphone
to only use opus.
I followed the opus install for asterisk and put the codec
in /usr/lib/asterisk/modules.
Looks like the opus codec is installed:
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@48000)->(slin@48000)->(slin@8000)->(ulaw@8000) opus:48000 To alaw:8000 : (opus@48000)->(slin@48000)->(slin@8000)->(alaw@8000) opus:48000 To gsm:8000 : (opus@48000)->(slin@48000)->(slin@8000)->(gsm@8000) opus:48000 To g726:8000 : (opus@48000)->(slin@48000)->(slin@8000)->(g726@8000) opus:48000 To g726aal2:8000 : (opus@48000)->(slin@48000)->(slin@8000)->(g726aal2@8000) opus:48000 To adpcm:8000 : (opus@48000)->(slin@48000)->(slin@8000)->(adpcm@8000) opus:48000 To slin:8000 : (opus@48000)->(slin@48000)->(slin@8000) opus:48000 To slin:12000 : (opus@48000)->(slin@48000)->(slin@12000) opus:48000 To slin:16000 : (opus@48000)->(slin@48000)->(slin@16000) opus:48000 To slin:24000 : (opus@48000)->(slin@48000)->(slin@24000) opus:48000 To slin:32000 : (opus@48000)->(slin@48000)->(slin@32000) opus:48000 To slin:44100 : (opus@48000)->(slin@48000)->(slin@44100) opus:48000 To slin:48000 : (opus@48000)->(slin@48000) opus:48000 To slin:96000 : (opus@48000)->(slin@48000)->(slin@96000) opus:48000 To slin:192000 : (opus@48000)->(slin@48000)->(slin@192000) opus:48000 To lpc10:8000 : (opus@48000)->(slin@48000)->(slin@8000)->(lpc10@8000) opus:48000 To g729:8000 : No Translation Path opus:48000 To speex:8000 : (opus@48000)->(slin@48000)->(slin@8000)->(speex@8000) opus:48000 To speex:16000 : (opus@48000)->(slin@48000)->(slin@16000)->(speex@16000) opus:48000 To speex:32000 : (opus@48000)->(slin@48000)->(slin@32000)->(speex@32000) opus:48000 To ilbc:8000 : (opus@48000)->(slin@48000)->(slin@8000)->(ilbc@8000) opus:48000 To g722:16000 : (opus@48000)->(slin@48000)->(slin@16000)->(g722@16000) opus:48000 To siren7:16000 : No Translation Path opus:48000 To siren14:32000 : No Translation Path opus:48000 To testlaw:8000 : (opus@48000)->(slin@48000)->(slin@8000)->(testlaw@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 But when I call I call answers but I have no audio. What am
I not doing ?
Thanks,
Jerry
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