Hi guys! Someone can share a opus configuration for webrtc thanks (Asterisk 17 version)!
I see sometimes syntax like: allow=opus:20 - i can’t find in documentation that is it (:20)?
Can i use 3 opus codecs in allow?
Thanks in advance
What exactly are you trying to accomplish in the end?
I want to make more quality calls from my crm system, through asterisk even if i have a not very quality connection, and somehow allow to swap to different codecs in this case (different configuration of opus may be)?
Changing the opus configuration is unlikely to alter that. It defaults to the best it can be, and things automatically adjust within the opus codec itself to a certain degree. What’s your scenario? Do other codecs have the same problem?
Sometimes i have this in logs:
[Jul 8 09:41:43] ERROR[7145][C-0000005c] codec_opus.c: Opus: decoding: corrupted stream
[Jul 8 09:41:43] ERROR[7145][C-0000005c] codec_opus.c: Opus: decoding: corrupted stream
[Jul 8 09:41:43] ERROR[7145][C-0000005c] codec_opus.c: Opus: decoding: corrupted stream
[Jul 8 09:41:43] ERROR[7145][C-0000005c] codec_opus.c: Opus: Unable to parse packet for number of samples: corrupted stream
[Jul 8 09:41:43] WARNING[7145][C-0000005c] translate.c: no samples for opustolin
[Jul 8 09:41:43] ERROR[7145][C-0000005c] codec_opus.c: Opus: decoding: corrupted stream
[Jul 8 09:41:43] ERROR[7145][C-0000005c] codec_opus.c: Opus: decoding: corrupted stream
[Jul 8 09:41:43] ERROR[7145][C-0000005c] codec_opus.c: Opus: Unable to parse packet for number of samples: corrupted stream
[Jul 8 09:41:43] WARNING[7145][C-0000005c] translate.c: no samples for opustolin
I don’t know why this happens, and want to understand is it needed to be somehow changed or not
No, configuration wouldn’t solve that issue. Which version of 17? What is heard when that occurs? Do other codecs work?
Yes 17 version. It’s a very poor quality of conversation. I don’t know how to check, this happens 1-2 times a day on 10 users. I think other configurations can solve this issue
There have been a few releases of 17, knowing which one is important as things will have been fixed. If you disable opus and use g722 instead, does that resolve the issue?
Connected to Asterisk 17.5.1 currently running on
I can try tomorrow to use g722 instead
i should just place allow=g722 instead of allow=opus?
Yes, that will enable g722 instead.
should i place g722 to voIP provider or only to webrtc side?
Ultimately, the provider will only give you access to a Mu-Law or A-law PSTN connection, so there is unlikely to be any quality advantage.
Is there any possibilities to increase quality on the voIP provider site (error corrections/dialplan options/echo cancellation or something like this)?
Bit error rates on PSTN are negligible.
The best place for echo cancellation (which is largely codec independent, and will favour simple codecs, is at the boundary between ISDN and VoIP, i.e. within your provider.
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