[asterisk-users] Choppy audio

[Eric "ManxPower" Wieling]

Make the card stop sharing it's IRQ with your IDE controller.  Try 
moving the card to another slot.

Asterisk has to send an audio packet every 20ms for VoIP calls.  I 
believe Zaptel expects no more than a few ms of latency.  If something 
is causing a delay, like the IDE controller locking interrupts and doing 
disk activity then you're not going to get your interrupts serviced fast 
enough and you will have audio issues.

Doug Crompton wrote:
> I am not sure who all see's this list but I do have a few questions that
> probably only the developers or somone really in the know of Asterisk
> could answer.
> 
> - What is the requirement for timing vs. audio playback in Asterisk.
> Specifically voicemail and IVR's (Not meetme)
> 
> - Has this requirment changed in newer versions?
> 
> This obviously is when using Asterisk with no internal cards. I used
> Asterisk for several years with a P3 Linux system, NO timing, and it
> worked flawlessly. Now with this new Pentium Dual core system I do not
> have the perfect audio I experienced with the less powerful system.
> 
> I fully know there are MANY variable here. It could be a combination of
> many things, including the OS (Linux Kernel) etc. BUT I offer this input,
> Music on Hold works fine. This uses mpg123. So why can this palyback fine
> and the other wav/gsm audio be choppy?
> 
> I would gladly switch to a newer Asterisk (using 1.2.29) if someone said
> this was solved in that version.
> 
> My system can obviously play (mpg123 - background) audio fine. Why then
> does Asterisk internal audio not also play well?
> 
> Doug
> 
> ****************************
> *  Doug Crompton	   *
> *  Richboro, PA 18954	   *
> *  215-431-6307		   *
> *		  	   *
> * doug at crompton.com        *
> * http://www.crompton.com  *
> ****************************
> 
> 
> 
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