Yealink Forums

I have several T21P E2 and T40P phones with latest firmware 82.0.20, Asterisk 14 (FreePBX 14 actually, installed with Asterisk 13, upgraded to Asterisk 14 with asterisk-version-switch) and lot of W56 handset (no Opus support yet). I use Asterisk PJSIP driver for my Yealink phones.
Opus was enabled everywhere and also set as primary codec. Asterisk console show calls established using opus (with opus->slin and slin->opus conversion actually).
In phones Opus enabled with autoprovision:

But voice sound as ... corrupted? Some scratches and crackles, you know.
First I blamed bad network connection, but it was not true - even phones in same LAN have this issue.
I tried to update Opus - download latest codec from Digium and install it to Asterisk. No luck, voice still bad.
I tried to disable Opus in PBX and replace it with G722 - sound become much better!

Should i stay with G722 for now or die trying try to solve this issue somehow?

(12-08-2017 10:56 AM)TrK Wrote: [ -> ]Asterisk console show calls established using opus (with opus->slin and slin->opus conversion actually).

Well there's your first problem. If you are calling two devices with OPUS enabled, Asterisk should be passing the codec through directly and not doing any type of conversion.

The fact that it is doing Opus - slin then slin -> Opus indicates that Asterisk is trying to process audio on the call for some reason. Now if you have installed everything properly and system is all running fine then you should not have any issues doing that, but I would suggest doing some digging on your PBX side to find out why the call is not simply being passed through as that may be the bigger point of the issue. (i.e. if you have call recording enabled on the PBX, then this will happen and maybe your PBX is not powerful enough to do both the recording to disk and the 2 x OPUS translation? That's just one possibility of many...)

(12-08-2017 11:05 PM)jolouis Wrote: [ -> ]

(12-08-2017 10:56 AM)TrK Wrote: [ -> ]Asterisk console show calls established using opus (with opus->slin and slin->opus conversion actually).

Well there's your first problem. If you are calling two devices with OPUS enabled, Asterisk should be passing the codec through directly and not doing any type of conversion.

You right, this is not good.
I was able to configure FreePBX to NOT do any converting with opus2opus calls, but this problem still hit me - sometimes call quality drops when using only opus and most calls have bad quality when opus converted to something other codec (even G722).
Now i use G722 only and feel good with this codec.

asterisk's opus sucks. badly. I believe they've got complexity turned WAY down and are using some old code. I never got satisfactory opus out of asterisk transcoding. Freeswitch does it great, as does 3CX.