[Asterisk-Dev] Help Debugging Dropped Call Audio

[Mike Benoit]

I got my hands on a couple of the raw .wav files, and it seems they do
not contain the artifacts I described in earlier emails. I ran my mp3
conversion script on the .wav's and the resulting mp3's do have the
artifacts (if played in XMMS only). So unless the MP3 conversion is
somehow picking up dropped audio packets which XMMS is exposing, this
doesn't appear to be the same issue you are running in to.

On Mon, 2005-12-26 at 17:12 -0500, Matthew Roth wrote:
> 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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-- 
Mike Benoit <ipso at snappymail.ca>
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