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.
Matt Roth
2005-Dec-21 15:31 UTC
[Asterisk-Users] Help Debugging Dropped Call Audio - Add'l Info
List users, I have some additional information related to the dropped audio. As always, I'd appreciate any help interpreting it. I set up an extension that calls the Milliwatt() application and is digitally recorded by the Monitor() application. Calls placed to it don't result in any of the custom warning messages and produce recordings free of skips/pops. I know that the same code is being executed, because it is also where the digital recordings are streamed to file. Therefore, it can be concluded that the problem lies somewhere in the differences between a SIP to SIP call and a SIP to Asterisk (in this case, the Milliwatt() application) call. The most obvious of the differences is that a SIP to SIP call involves two active SIP channels that are bridged together. A SIP to Asterisk call only involves one active SIP channel. I confirmed this by placing calls of both types and observing the output of the "sip show channels" command in the CLI. To confirm that it is not simply a matter of the number of active SIP channels on the Asterisk server, I placed two simultaneous SIP to Asterisk calls and observed that no custom warning messages were generated. On a SIP to SIP call, a custom warning message is almost always generated from the ast_read() function as soon as the callee picks up the phone and the SIP channels are bridged. The messages then appear sporadically throughout the call, and there is often a final message generated when the call is terminated. Keep in mind, each of these messages is indicative of a short drop in the call audio which can be heard as a pop/click on the PCM format recording. All of this leads me to believe that the problem dwells somewhere in the code responsible for bridging the channels. Does anyone have any ideas as to the specific cause or the direction that I should go to further pursue it? I've heard that Asterisk's locking scheme is questionable, but I don't know if that would apply to this scenario. I'd still like to hear if anyone else has replicated the problem, as well. Thanks, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer 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-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >