Most calls are at 8kHz. The formula for bandwidth is sampling rate * bit rate * channels, so 8000 * 8 * 1 = 64000 = 64kbps (per way).

Math

On 18-Jun-09, at 1:01 PM, Andy wrote:

Thanks Brian,
 
So, just to calrify will the base call always be 8kHz?
 
On a related note, do you happen to know the bitrate of each open channel/live call? Is it 16 kilobits per second like the recorded audio? I need to do some calculations on the badwidth required to handle a certain number of concurrent calls.
 
Many thanks
Andy


From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West
Sent: 18 June 2009 18:11
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Sample rate and recordFile


On Jun 18, 2009, at 11:54 AM, Andy wrote:

1) I notice that when I change the sample rate it automatically changes the bit rate too. I understand why this is the case but wondered if it was just as easy to be able to control the bitrate as well as the sample rate.

If you're talking about mod_shout, NO.  You'll end up picking an invalid bitrate and asking why it doesn't work... been there done that... I changed it a few months back to pick the optimal bitrate for the sample rate.

2) When I use a sample rate other than 8000 I get a warning 'Sample rate doesn't match'. I guess this puts some extra load on the server. If all my calls are being recorded and all at 11025 can/should I alter the sample rate of the base call to 11025?

NO.  Your phone call is running at 8kHz, Your sound file is 11025 and they don't match, If you were to play this file into an 8k channel without a resample it would sound a little like satan.  or a dragging tape deck.  The file has to be resampled to match the current session rate. 

/b

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