[Asterisk-Dev] Help Debugging Dropped Call Audio

Matthew Roth mroth_imm at hotmail.com
Mon Dec 26 15:12:17 MST 2005


Mike Benoit wrote:

>On Mon, 2005-12-26 at 13:40 -0500, Matthew Roth wrote:
> > Mike Benoit wrote:
> > - What have you tried to solve the problem (ie. Disabling the Linux 
>frame
> > buffer, not running X windows, disabling hyperthreading, disabling ACPI,
> > etc.)?
>
>Haven't tried anything, like I said, I just use SOX's "play" application
>and it seems good enough. We record all our calls, but rarely ever
>listen to them.

Out of curiosity, are you running X windows on the server or using the Linux 
frame buffer?  Both are known to cause jittery sound.

http://www.voip-info.org/wiki/index.php?page=Asterisk+X11
http://www.voip-info.org/wiki/index.php?page=Asterisk+disable+frame+buffer

Is your Zap hardware sharing interrupts?

http://www.voip-info.org/wiki/view/Asterisk+hardware+interrupts

I'm trying to eliminate other possibilities to ensure that we're 
experiencing the same issue.

> > - What codec are you using for the calls?
>
>Usually its ULAW -> ZAP if the call is going out a local trunk. Or ULAW
>-> GSM if it goes out a VOIP provider. It seems to happen in all cases.

That's an interesting piece of information.  Thanks.

> > - What format are the calls being recorded to?
>
>Monitor(wav,${TIMESTAMP}-${CALLERIDNUM}-${MACRO_EXTEN}, m), then we mix
>the files and convert them to MP3's in the evening.

If I'm understanding this correctly, at call completion you are left with a 
single WAV file that is the full recording of the call.  Then you run a 
nightly batch process to convert the WAVs to MP3s.  Is that correct?

>Unfortunately I don't have any of the recordings before they were
>converted to MP3's right now (I will in a couple days), its been so long
>since I've listened to them I can't recall if the original files have
>the cracks/pops or not. Do you want me to send you a couple of the MP3's
>anyways?

If it's possible, could I get some recordings prior to the conversion to 
MP3?  MP3 is pretty bad with voice, so it introduces a number of compression 
artifacts (mostly tinny sound).  I know my way around sox/soxmix pretty 
well, so the original format would be ideal.  Otherwise, I'd be happy to 
listen to a few of the MP3s.

> > - What call volumes do you experience the problem at?
>
>I haven't adjusted the volumes in quite a while, but it seems to happen
>in all calls.

Allow me to clarify.  By call volume, I meant the number of concurrent calls 
on the system.

> > - Are you recording the leg files directly to disk (we record to a RAM
> > disk)?
>
>Directly to disk. The server is very low usage (4 phones, 4 trunks)
>Rarely are there more then 2 phones in use at anytime.
>
> > - What is your hardware platform?
>
>model name      : AMD Athlon(tm) Processor
>cpu MHz         : 1145.685
>
>IDE harddrive.
>
> > - What Digium hardware are you using?
>
>We have a couple clone ZAP interfaces, which we use in "overflow"
>scenarios, and 2 SPA-3000's, and 2 SPA-2000's.
>
> > - What is your Linux distribution?
>
>Mandrake 10
>
> > - What is your kernel version?
>
>2.6.3-13mdk
>
>This is just one of the Asterisk boxes, the others are very similar
>though.
>
>--
>Mike Benoit <ipso at snappymail.ca>

Thanks for the information,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer





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