I am using MixMonitor to record the both sides of the conversation and it’s saved as a wav file. I then convert the wav file to mp3 to save disk space. The problem is the original wav file is 8kbit and 8000Hz Mono audio file which is a low quality audio. I am wondering if there is a way to record with a better quality in asterisk. Right now i am not concerned about the disk space or the format of the audio file (I can convert it to something else later).
For some reason I have this idea in my mind that all the phone lines are 8kbit and there is nothing we can do about it to improve the quality but I am not sure if i am mistaken or not.
Are there any suggestions?
Thanks,
PSTN lines are 8000 samples per second, using 8 bit companded from 16 bit (using G.711 A or mu-law) samples. 16 bit by 8000 samples per second wave files will exceed the quality achievable on digital PSTN systems. Some long distance connections, and all mobile phone ones, will be lower quality than this.
Historically the lines were 300 to 3.4kHz analogue, but the noise level would have been similar to or greater than the quantisation noise from G.711. (Historical systems used frequency division multiplexed 4kHz channels.)
Rather than worrying about the quality of the recording, you should be thinking or either reducing the MP3 bit rate to the minimum, or one step above the minimum, or using a speech optimised codec, like GSM.