[Asterisk-Dev] Help Debugging Dropped Call Audio

[Mike Benoit]

Matthew,

	If it helps, I have been dealing with this issue on about 5 different
Asterisk boxes for the last year or more. Early v1.0 all the way up to
v1.2. These boxes are all very low usage (couple lines max), but this
problem happens consistently in every recording made by Monitor. The
weird thing though is if we listen to the recording in XMMS, the
cracks/pops are terrible, and make it very difficult to listen to. But
if I use the SOX package's "play" application, the cracks/pops
disappear. I'm not sure if its just masking them, but recording sounds
much better, and I haven't NOTICED problems of dropped audio.

During the calls I can't determine that any audio is being dropped. 

Since using "play" or any other windows based media player that I have
tried doesn't have this crack/pop issue, I haven't reported this problem
to the mailing list at all.

Let me know if you would like more information.

PS. This happens with SIP->SIP calls, AND SIP->ZAP calls. 

On Wed, 2005-12-21 at 18:06 -0500, Matt Roth wrote:
> List users,
> 
> Below is a bug report documenting Asterisk dropping call audio at very
> low loads (1 call).  I have personally reproduced it on three separate
> machines, multiple network architectures (including a 48-port Cisco
> Catalyst 3560 POE switch dedicated to an Asterisk server and two Snom
> 320 VoIP phones), and three versions of Asterisk.
> 
> Despite this effort, I would still like to insure that I am not
> experiencing an isolated problem.  That is where I need your help.  If
> you could follow the steps in the bug report and post your results to
> the list, it would be greatly appreciated.
> 
> Thank you,
> 
> Matthew Roth
> InterMedia Marketing Solutions
> Software Engineer and Systems Developer
> 
> ================================================================================
> 
> Description of problem:
> Dropped audio during SIP to SIP, u-Law calls manifesting itself as
> clicks/pops in PCM format recordings.
> 
> Version-Release number of selected component (if applicable):
> A.2-beta, A.2-1, SVN-branch-1.2-r7231
> 
> How reproducible:
> Always
> 
> Steps to Reproduce:
> 1. Setup SIP to SIP calls to use the u-Law codec
> 2. Setup SIP to SIP calls to be recorded in PCM format via the Monitor
> application
> 3. Conduct a 5 to 10 minute SIP to SIP call
> 4. Mix the leg files with soxmix (soxmix -v 1.0 -t ul LEG-IN.PCM -t ul
> LEG-OUT.PCM -g -b MIXED.WAV)
> 5. Listen to the mixed file
> 
> Actual Results:
> Periods of dropped audio in the call can be heard as clicks/pops in the
> recording.
> 
> Expected Results:
> Clear call audio and recording.
> 
> Additional info:
> The format that the call is recorded in is relevant.  The PCM format
> accentuates the dropped audio as a click/pop, while the GSM format masks
> it as periods of (sometimes imperceptible) silence.  Therefore, it is
> extremely important to record in PCM format to diagnose the problem.
> 
> Adding an ast_log() call to the ast_read() and ast_write() functions in
> channel.c that logs calls to the ast_seekstream() function can be
> helpful in debugging the problem.
> 
> For example:
> 
>     /* From ast_read() */
>     int jump = chan->outsmpl - chan->insmpl - 2 * f->samples;
>     if (jump >= 0) {
>         if (ast_seekstream(chan->monitor->read_stream, jump +
> f->samples, SEEK_FORCECUR) == -1)
>             ast_log(LOG_WARNING, "Failed to perform seek in monitoring
> read stream, synchronization between the files may be broken\n");
>         chan->insmpl += jump + 2 * f->samples;
>         /* Log calls to ast_seekstream */
>         ast_log(LOG_WARNING, "Performed %d sample jump in monitoring
> read stream to synchronize the leg files\n", jump + f->samples);
>     } else
>         chan->insmpl+= f->samples;
> 
> All dropped call audio will now be accompanied by the "Warning"
> statement that has been added.  Note these as they appear in the console
> (or messages log) then listen to the recording.  You will see that the
> drops in call audio, the
> clicks/pops in the recording, and the warnings are occurring at
> precisely the same moment.
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> Asterisk-Dev mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-dev
-- 
Mike Benoit <ipso at snappymail.ca>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: application/pgp-signature
Size: 189 bytes
Desc: This is a digitally signed message part
Url : http://lists.digium.com/pipermail/asterisk-dev/attachments/20051225/83c2064f/attachment.pgp