Moh quality and best practice

Is there a best practice to import MoH files into freepbx ?
I usually load 16bitPCM wav (don’t care if stereo or 44KHz)
It will be converted by freepbx to chosen format(s).

Which format is better ? ulaw alaw wav etc. , why ?

Due to usually poor quality imposed by telco chain , should one emphasize low/high frequencies with common equalizer softwares while processing sound files ?

If you use ulaw codec then the ulaw format wont require transcoding. Wav to ulaw does.

What poor quality are you refering to?

The absolute best is one format each for all the codecs in use, because there is no transcoding cost and no possibility of degradation.

If the telco chain involves mobile phones, which only get a speech service, you will need to experiment with the choice of any music, as the system is not designed for music. If you are just dealing with landline, and therefore, at least, a 3.1kHz audio service pre-emphasis is unlikely to help and may produce aliasing artefacts by forcing through frequencies above the, nominal, 3.4kHz upper limit which may then get aliased to 4kHz - f.

There is a trend to higher audio bandwidth, but you want to optimise for 3.1kHz audio. Although this trend also applies to mobile networks, they still only provide a speech service.

Tom , I’m referring mostly to landline bandwidth limits and degradation.
Ok for the transcoding cost.
David I was right thinking to artefacts produced by “pumping” the audio band edges…
Perhaps the MoH file quality is the main factor to deal to.

Well if you are referring to TDM/copper in a traditional sense, they do ulaw/alaw mainly with some g729. Yup, codecs are codecs in that manner. So the old copper landline doing ulaw is doing the same ulaw as your SIP line except that the SIP line is using more bandwidth due to IP overhead.

Also, keep in mind that the codecs being used between you and your provider are between you and your provider. Just because your provider allows you to use g722 with them (for example) doesn’t mean the other side (Caller or callee) is doing g722. You can’t control one half of this PSTN call and what codecs they use. Your side can use g722 or Opus but if the other side is still forcing g729 to their customers…they’re going to transcode it as such.

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