CP920 audio quality issues

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newuser1

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Hello,

I was wondering if any one else is experiencing audio quality issues when using this model Yealink phone? We have tried replacing these phones with new ones, but we get reports that the call audio will intermittently be garbled. We have numerous T42S phones at the site that don't experience this issue. Could it be their is a bug in the firmware? Phones are running firmware 78.84.0.95.
 
While it could be a number of things, the prime suspect, when it comes to audio problems, is your choice of Codecs. Check the Activity Log to see which ones are being used, and if any transcoding is happening. You don't say if this is internal, external, what your network set-up is...
 
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PCMU, PCMA, G722, G729 are the codecs being used top down
 
The issue was reported from upper management. I couldn't replicate the issue with a external local call.
 
You'd need to be able to replicate the issue, to start to narrow down the cause.
 
Yeah, I took a capture on our switch and 3cx server. I plan on opening a case with 3cx support. THanks for the guidance.
 
I don't think there is really an issue. Ultimately issues on conference calls are going to be more noticeable because there are more participants and they run longer, increasing the possibility of either an outside caller's poor quality being heard and assumed to be a local issue or an actual intermittent local issue. Keep in mind that internet in general is working harder right now that it used to so it could simply be a strain on your individual connection, regional network, etc.
 
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I imagine the garbled audio was heard on the CP920, meaning in may have come in to your trunk garbled.

If you have recordings enabled, and the above is true, then the recording will also contain garbled audio. A good place to check when you cannot replicate it yourself.
 
Is there a call flow diagram for external calls?

My understanding is
Desk phone <=> local SBC <=> 3CX server <=> SIP Trunk <=> "Cloud" <=> Telco <=> Cell Phone

I have opened a case with my sip provider and they said they only handle sip handling and audio is not delivered by them, is this a true statement?
 
Your understanding is generally correct.

As for the audio, it has to be sent to your PBX from the provider, and it will pass it back down the same flow in the opposite direction.

Audio from the other side is delivered from your provider back to your PBX, and from there they no longer control what happens to the audio. Perhaps they meant audio is not delivered by them to the endpoint (which would be true, the PBX is going to deliver that audio depending on how you are set up)

Edit: if they indeed do not send audio, then I suggest you change provider to a supported one
 
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