I’m new to FreePBX and couldn’t find a procedure for my problem.
I like to ask you how to achieve best sound quality for my recordings and MoH.
I produce them my own and have kind of advanced recording equipment (Rode-NT1, Steinberg uv-44, Cubase), meaning I could export my recordings (piano, voice, vocals etc.) to whatever quality.
I’m using FreePBX 126.96.36.199
My SIP Provider supports G.722, ulaw and alaw
My Phones all support the same codecs.
What format, bitrate, khz would you choose to get best quality?
For outside calls: What I understand is that for calls my provider and the callers provider look for a codec that matches. If the callers provider may not support G.722 is the higher Quality automatically converted to lower (ulaw, alaw) depending on their providers quality?
How would you proceed to make it sound best?
Thank you for your help,
Hi @Tiehscher !
You can use g722, alaw or ulaw, but you need to know that quality is not going to be “good” for outside callers, telephony is mono by definition, PSTN prioritizes voice over music, so it won’t support stereo.
Thank you @slobera for your answer. So of course I mixdown the recording to mono. But how many bit and kHz would you propose and what file format should I use?
You can record your files with the quality you want, I use to work with this page: https://g711.org/ which allows you to convert your file to an Asterisk supported file and also gives you the chance to optimize the audio for phones (using the bandpass filter) and select the volume.
Thanks. I’ll try that. That sound very promising!
And when you upload the files, you use the GUI to do it and make sure you select G722, ulaw, and alaw to convert it to when you add it.
That way the PBX is doing as little live transcoding as possible.
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