Does Third lane support "Wideband" audio...is PBX Manager Version 6.1.1.8 compatible?
I am using PBX Manager Version 6.1.1.8 and all of my phones are manufactured by Yealink T20 T22 T26 T28 etc. Is there any way I can get around from using low quality 8 KHz audio on our PBX? I am not satisfied with 8 KHz recordings. I come from the world of video where 48 KHz and 96 kHz are the norm and my ears can really tell the difference.
Sample Rate "eh Ok"
I found that you can put a band aid on 8khz recordings with a little audio sweetening and get "OK" results. However it still seems like this technology is still in the stone age if 8Khz is the best we can do.
1.) Record at 192Khz resolution
2.) Graphic EQ 300hz - 4khz frequency bands. Bring down all other frequencies outside of this frequency band 6db (This makes the sound like a higher sample rate when it is not) 300hz is around the muddy frequencies that typically sound like mud.
3.) Normalize
4.) Compressor (Speech)
5.) Declicker
6.) Waves Max Bass
7.) Resample to 8Khz
8.) Save
the entire pstn is 8k, the
the entire pstn is 8k, the higher the sample rate the higher the bandwidth.. and bandwidth boils down to money. When youre using 8k payloads with g729 compression youre looking at 8kbps payloads and 29kbps bandwidth. Even ulaw is 64kbps payload and 80kbps bandwidth. A provider sending you calls from the pstn needs to be able to handle at least 100,000 concurrent calls. At g711,ulaw thats 7 gigabit of bandwidth; at g729 thats 3 gigabit. There just inst an infrastructure in 192khz sample rates, with the best possible compression thats 220kbps for a single call.
i think you should do some reading, starting with "Asterisk - The future of Telephony". You really think your IVR is going to sound any different with a 16k sample rate when someone from the outside calls in? Thats like arguing about printing something in hi-rez on quality photo paper right before faxing it to the other side. The only person who will benefit from an IVR using something like g722 is going to be your yealink phone, and why the hell are you calling your own ivr in the first place? Noone on the outside is going to hear anything but an 8k sample. In fact, asterisk now has to resample and transcode your audio file just to play it to the outside world. The net result is the audio is worse than even 8k recorded files, AND you've adding load to your pbx to do this on-the-fly constant transcoding.