or go over the limit and you'll have Max Headroom =D<br><br><br><div class="gmail_quote">On Thu, Jun 18, 2009 at 1:16 PM, David Knell <span dir="ltr"><<a href="mailto:dave@3c.co.uk">dave@3c.co.uk</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">- plus UDP/RTP overhead. Budget 10 calls/megabit for G.711 and you'll<br>
have a bit of headroom available.<br>
<br>
--Dave<br>
<div><div></div><div class="h5"><br>
> Most calls are at 8kHz. The formula for bandwidth is sampling rate *<br>
> bit rate * channels, so 8000 * 8 * 1 = 64000 = 64kbps (per way).<br>
><br>
><br>
> Math<br>
><br>
> On 18-Jun-09, at 1:01 PM, Andy wrote:<br>
><br>
> > Thanks Brian,<br>
> ><br>
> > So, just to calrify will the base call always be 8kHz?<br>
> ><br>
> > On a related note, do you happen to know the bitrate of each open<br>
> > channel/live call? Is it 16 kilobits per second like the recorded<br>
> > audio? I need to do some calculations on the badwidth required to<br>
> > handle a certain number of concurrent calls.<br>
> ><br>
> > Many thanks<br>
> > Andy<br>
> ><br>
> ><br>
> > ____________________________________________________________________<br>
> > From: <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a><br>
> > [mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a>] On Behalf Of<br>
> > Brian West<br>
> > Sent: 18 June 2009 18:11<br>
> > To: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
> > Subject: Re: [Freeswitch-users] Sample rate and recordFile<br>
> ><br>
> ><br>
> ><br>
> ><br>
> > On Jun 18, 2009, at 11:54 AM, Andy wrote:<br>
> ><br>
> > > 1) I notice that when I change the sample rate it automatically<br>
> > > changes the bit rate too. I understand why this is the case but<br>
> > > wondered if it was just as easy to be able to control the bitrate<br>
> > > as well as the sample rate.<br>
> ><br>
> ><br>
> > If you're talking about mod_shout, NO. You'll end up picking an<br>
> > invalid bitrate and asking why it doesn't work... been there done<br>
> > that... I changed it a few months back to pick the optimal bitrate<br>
> > for the sample rate.<br>
> ><br>
> > > 2) When I use a sample rate other than 8000 I get a warning<br>
> > > 'Sample rate doesn't match'. I guess this puts some extra load on<br>
> > > the server. If all my calls are being recorded and all at 11025<br>
> > > can/should I alter the sample rate of the base call to 11025?<br>
> ><br>
> > NO. Your phone call is running at 8kHz, Your sound file is 11025<br>
> > and they don't match, If you were to play this file into an 8k<br>
> > channel without a resample it would sound a little like satan. or a<br>
> > dragging tape deck. The file has to be resampled to match the<br>
> > current session rate.<br>
> ><br>
> ><br>
> > /b<br>
> ><br>
> ><br>
> > _______________________________________________<br>
> > Freeswitch-users mailing list<br>
> > <a href="mailto:Freeswitch-users@lists.freeswitch.org">Freeswitch-users@lists.freeswitch.org</a><br>
> > <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
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><br>
><br>
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</div></div><font color="#888888">--<br>
David Knell, Director, 3C Limited<br>
T: +44 20 3298 2000<br>
E: <a href="mailto:dave@3c.co.uk">dave@3c.co.uk</a><br>
W: <a href="http://www.3c.co.uk" target="_blank">http://www.3c.co.uk</a><br>
</font><div><div></div><div class="h5"><br>
<br>
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