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Summary:ASTERISK-11533: Listening to Allison voicemail prompt on SIP phone causes [pop] sounds
Reporter:mustardman (mustardman)Labels:
Date Opened:2008-02-27 17:57:02.000-0600Date Closed:2008-03-27 12:13:06
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Sounds
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) after_padulaw.zip
( 1) popping_sounds.zip
Description:This is something that has been bugging me for awhile.  I have noticed it on multiple systems with different hardware, different brands and models of hardphones, with and without zaptel cards, and various versions of Asterisk in the 1.2 and 1.4 branches.

A bit hard to describe.  Using a SIP hardphone I log into my voicemail at which point Allison says "you have x messages.....".  There are various other prompts that exhibit the same problem that are not necessarily part of voicemail but that is one easy to explain and reproduce one. The problem is there is a slight 'pop' sound usually during the first syllable of each word.  So the prompt sounds like "y[pop]ou h[pop]ave t[pop]wo me[pop]ssages".  If I dial *43 to do an echo test and Allison says "you are about to enter an echo test....." it's not there so it's only in certain modes this happens.

It's relatively minor and not something people generally complain about but it's definitely there and (at least to me) kinda makes the system seem to be a bit less polished sounding.  I thought it had something to do with the prompts I am using but when I play back the actual files in a sound player I don't hear those pops.  I tried using various format prompts including gsm, ulaw etc. but it doesn't change anything.

This problem has always been there for as long as I can remember.  Doesn't matter what version Asterisk/Zaptel/Sounds I am using.

Comments:By: Jason Parker (jparker) 2008-03-11 14:38:15

We need a great deal more information in order to even begin debugging something like this.
By: mustardman (mustardman) 2008-03-11 14:46:26

If you have an asterisk system nearby with a SIP phone you will be able to reproduce it regardless of the what hardware and what versions of software you are using.  So are you saying you cannot reproduce it or that you would like me to supply you with the necessary information?  If the latter just tell me what you need.
By: Jason Parker (jparker) 2008-03-11 15:03:04

This is not something I'm able to reproduce.  What codec prompts are you using?
By: mustardman (mustardman) 2008-03-11 18:02:05

I am using the currently using the basic and additional ulaw sounds downloaded from the release branch on the asterisk website.  I seem to recall noticing the same issue when I was using gsm.  The phones and Asterisk all run the default G711u codec.  I am also running the FreePBX GUI.

I just tried dialing into DISA from a landline and accessing voicemail.  The artifacts are still there but not nearly as noticeable as when doing it from a SIP extension.  They sound more like a slight warble in her voice rather than the "POP' sound when listening from a SIP phone.  Probably because of the quality difference.


By: mustardman (mustardman) 2008-03-11 18:40:26

Xlite has a record function so I was able to record the actual problem which I have uploaded.  I hear the same popping sounds on Aastra hard phones so it is independent of the SIP phone used.

After I enter the mailbox you can clearly hear the slight pop/warble in Allisons voice on the first syllable of each sound file when she says "you have no messages".

Curiously, I don't hear any popping sounds when she is prompting me for mailbox and password.


By: Andrew Lindh (andrew) 2008-03-12 15:26:14

I have the same problem with VM and other prompts too.... I'm using G.711 and only SIP phones/gateways (polycom and cisco) and I have a digium card in the server for timing.
By: Doug Lytle (lytledd) 2008-03-15 08:22:23

Could this be related to the GCC 'Don't compile with Optimize' issue?

Try using the compiler flag of 'Don't optimize', recompile and see if that helps
By: Andrew Lindh (andrew) 2008-03-17 16:17:01

DONT_OPTIMIZE did not change anything for me.... Voice calls are always good, it's just prompt playback from the system. zttest is better than 99.995% all the time.


By: Jason Parker (jparker) 2008-03-18 13:17:26

I listened to the file that you've supplied, and it just sounds like it was transcoded multiple times.  I also listened to the stock gsm prompts, and they do have some slight issues like this..  I'm fairly certain that this is just a result of over-transcoding, and not something Asterisk is specifically doing.

I'm going to leave this open for a short while, but I feel that there is nothing we can really do to resolve this, except for you to use better quality sounds (not gsm), and don't transcode unless needed.

You should run `make menuselect` and install the ulaw prompts.  You'll also need to verify that you aren't transcoding back and forth.  That should help a little bit, at least.
By: Andrew Lindh (andrew) 2008-03-18 15:19:49

What is the primary version of the Allison prompts? WAV?
By: mustardman (mustardman) 2008-03-20 21:24:47

I recompiled with latest (v1.4.18.1) asterisk after make menuselect and unchecking everything except ulaw in core and addon sounds.  I made sure I only had *.ulaw files in /var/lib/asterisk/sounds and all subdirectories.  I set my phones to ONLY use ulaw.

sip.conf has:
disallow=all
allow=alaw
allow=ulaw

at CLI>sip show channels
shows that I am running ulaw during call

Nothing has changed.  I still get the same artifacts!


By: Jason Parker (jparker) 2008-03-21 10:50:42

Are you recording all calls?  That would be transcoding to slin and back...
By: BicomSystems Ltd. (fkasumovic) 2008-03-21 11:16:54

Try this:
http://threebit.net/mail-archive/asterisk-users/msg28844.html

Compile that application and run it like this
find /var/lib/asterisk/sounds -type f -name '*.ulaw' | xargs padulaw

For alaw files, just change ULAW_SILENCE 0xff with 0x55, and apply to .alaw files.
By: Andrew Lindh (andrew) 2008-03-21 14:47:29

Thanks for the pointer.... I had not thought of that issue as I assumed asterisk would be able to deal with audio that was not frame aligned! I tried this with the sound file "zip-code" (no transcoding, ulaw file, ulaw SIP phone). When I play the standard version 4 times in a row and I hear pops, if I play the padded version I don't hear pops any more. I guess it's not a defect in the file, it's a defect somewhere in asterisk with how it plays back audio files. I don't know if it's sending short frames and the phone is not doing a good job of PLC or if it's asterisk sending bad audio data. I tested this using Asterisk 1.6.0
By: BicomSystems Ltd. (fkasumovic) 2008-03-24 01:12:15

Notice that it dose not happen when asterisk plays single file.
It happens when it plays multiple files, like voicemail application.
You will probably find out that first time it say password there is no 'pop' sound. If you enter wrong password , you will notice next time it say password 'pop' sound somewhere 's' and between 'r' and 'd'.
By: mustardman (mustardman) 2008-03-25 18:12:39

padulaw fixes it.  I have uploaded the same voicemail session after running sounds through padulaw.  Here is the code just to make sure it's documented.  Perhaps Digium could incorporate something like this on the Asterisk playback end or at the very least, process their sound files.  It's definitely a nice refinement.

/* padulaw.c */

>SNIP<


By: Jason Parker (jparker) 2008-03-25 18:39:45

Please do not post code (especially code you do not have copyright to) in comments.
By: mustardman (mustardman) 2008-03-25 19:17:04

.


By: Digium Subversion (svnbot) 2008-03-26 18:22:14

Repository: asterisk
Revision: 111245

U   branches/1.4/main/frame.c

------------------------------------------------------------------------
r111245 | qwell | 2008-03-26 18:22:12 -0500 (Wed, 26 Mar 2008) | 9 lines

Remove excessive smoother optimization that was causing audio glitches (small "pops")
after (about 200ms later) an "incorrectly" sized frame was received.

While it would be very nice to keep this as optimized as possible, it makes no sense
for the smoother to be dropping random bits of audio like this.  Isn't that the
whole point of a smoother?

Closes issue ASTERISK-11533.

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=111245
By: Digium Subversion (svnbot) 2008-03-26 18:23:15

Repository: asterisk
Revision: 111246

_U  trunk/
U   trunk/main/frame.c

------------------------------------------------------------------------
r111246 | qwell | 2008-03-26 18:23:14 -0500 (Wed, 26 Mar 2008) | 17 lines

Merged revisions 111245 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r111245 | qwell | 2008-03-26 18:26:33 -0500 (Wed, 26 Mar 2008) | 9 lines

Remove excessive smoother optimization that was causing audio glitches (small "pops")
after (about 200ms later) an "incorrectly" sized frame was received.

While it would be very nice to keep this as optimized as possible, it makes no sense
for the smoother to be dropping random bits of audio like this.  Isn't that the
whole point of a smoother?

Closes issue ASTERISK-11533.

........

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=111246
By: Digium Subversion (svnbot) 2008-03-26 18:23:36

Repository: asterisk
Revision: 111247

_U  branches/1.6.0/
U   branches/1.6.0/main/frame.c

------------------------------------------------------------------------
r111247 | qwell | 2008-03-26 18:23:35 -0500 (Wed, 26 Mar 2008) | 25 lines

Merged revisions 111246 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
r111246 | qwell | 2008-03-26 18:27:33 -0500 (Wed, 26 Mar 2008) | 17 lines

Merged revisions 111245 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r111245 | qwell | 2008-03-26 18:26:33 -0500 (Wed, 26 Mar 2008) | 9 lines

Remove excessive smoother optimization that was causing audio glitches (small "pops")
after (about 200ms later) an "incorrectly" sized frame was received.

While it would be very nice to keep this as optimized as possible, it makes no sense
for the smoother to be dropping random bits of audio like this.  Isn't that the
whole point of a smoother?

Closes issue ASTERISK-11533.

........

................

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=111247
By: Digium Subversion (svnbot) 2008-03-27 12:13:06

Repository: asterisk
Revision: 111440

_U  team/murf/bug11210/
U   team/murf/bug11210/UPGRADE.txt
U   team/murf/bug11210/apps/app_playback.c
U   team/murf/bug11210/apps/app_voicemail.c
U   team/murf/bug11210/channels/chan_sip.c
U   team/murf/bug11210/channels/misdn_config.c
U   team/murf/bug11210/contrib/scripts/autosupport
U   team/murf/bug11210/include/asterisk/ael_structs.h
U   team/murf/bug11210/include/asterisk/linkedlists.h
U   team/murf/bug11210/include/asterisk/pval.h
U   team/murf/bug11210/main/config.c
U   team/murf/bug11210/main/frame.c
U   team/murf/bug11210/main/http.c
U   team/murf/bug11210/main/pbx.c
U   team/murf/bug11210/pbx/pbx_ael.c
U   team/murf/bug11210/res/ael/pval.c

------------------------------------------------------------------------
r111440 | murf | 2008-03-27 12:13:02 -0500 (Thu, 27 Mar 2008) | 182 lines

Merged revisions 111067,111083,111123,111127,111130,111132,111185,111213,111246,111285,111295,111360,111410 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r111067 | mmichelson | 2008-03-26 13:26:23 -0600 (Wed, 26 Mar 2008) | 17 lines

Merged revisions 111049 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r111049 | mmichelson | 2008-03-26 14:22:16 -0500 (Wed, 26 Mar 2008) | 9 lines

Add a lock to the vm_state structure and use the lock around mail_open calls
to prevent concurrent access of the same mailstream. This, along with trunk's
ability to configure TCP timeouts for IMAP storage will help to prevent
crashes and hangs when using voicemail with IMAP storage.

(closes issue ASTERISK-10111)
Reported by: ewilhelmsen


........

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r111083 | file | 2008-03-26 13:29:26 -0600 (Wed, 26 Mar 2008) | 4 lines

Add expiry value to the sip show subscriptions CLI command.
(closes issue ASTERISK-11470)
Reported by: agx

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r111123 | mmichelson | 2008-03-26 13:39:23 -0600 (Wed, 26 Mar 2008) | 12 lines

Merged revisions 111121 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r111121 | mmichelson | 2008-03-26 14:37:36 -0500 (Wed, 26 Mar 2008) | 4 lines

This code change is made just for clarification. It does exactly
the same thing as before. It just doesn't look as wrong.


........

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r111127 | kpfleming | 2008-03-26 13:52:27 -0600 (Wed, 26 Mar 2008) | 18 lines

Merged revisions 111126 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r111126 | kpfleming | 2008-03-26 14:51:24 -0500 (Wed, 26 Mar 2008) | 10 lines

Merged revisions 111125 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r111125 | kpfleming | 2008-03-26 14:49:30 -0500 (Wed, 26 Mar 2008) | 2 lines

update UPGRADE notes to document usage of the script

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r111130 | file | 2008-03-26 13:56:40 -0600 (Wed, 26 Mar 2008) | 14 lines

Merged revisions 111129 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r111129 | file | 2008-03-26 16:55:08 -0300 (Wed, 26 Mar 2008) | 6 lines

Update autosupport script.
(closes issue ASTERISK-10279)
Reported by: angler
Patches:
     autosupport.diff uploaded by angler (license 106)

........

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r111132 | tilghman | 2008-03-26 13:58:09 -0600 (Wed, 26 Mar 2008) | 2 lines

Simplify new macro, simplify configfile logic, now that list is sorted

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r111185 | tilghman | 2008-03-26 14:34:05 -0600 (Wed, 26 Mar 2008) | 2 lines

Oops, missed one

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r111213 | twilson | 2008-03-26 15:23:29 -0600 (Wed, 26 Mar 2008) | 2 lines

Stupid strcasecmp function :-)

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r111246 | qwell | 2008-03-26 17:27:33 -0600 (Wed, 26 Mar 2008) | 17 lines

Merged revisions 111245 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r111245 | qwell | 2008-03-26 18:26:33 -0500 (Wed, 26 Mar 2008) | 9 lines

Remove excessive smoother optimization that was causing audio glitches (small "pops")
after (about 200ms later) an "incorrectly" sized frame was received.

While it would be very nice to keep this as optimized as possible, it makes no sense
for the smoother to be dropping random bits of audio like this.  Isn't that the
whole point of a smoother?

Closes issue ASTERISK-11533.

........

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r111285 | qwell | 2008-03-26 18:25:56 -0600 (Wed, 26 Mar 2008) | 9 lines

Merged revisions 111280 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r111280 | qwell | 2008-03-26 19:25:13 -0500 (Wed, 26 Mar 2008) | 1 line

Put this flag back so we don't change the API.
........

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r111295 | qwell | 2008-03-26 18:27:35 -0600 (Wed, 26 Mar 2008) | 1 line

But we can change the API here.
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r111360 | murf | 2008-03-26 22:47:12 -0600 (Wed, 26 Mar 2008) | 23 lines

Merged revisions 111341 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r111341 | murf | 2008-03-26 21:21:05 -0600 (Wed, 26 Mar 2008) | 15 lines


(closes issue ASTERISK-11725)
Reported by: pj
Tested by: murf

These changes will set a channel variable ~~EXTEN~~ just before generating code
for a switch, with the value of ${EXTEN}. The exten is marked as having a switch,
and ever after that, till the end of the exten, we substitute any ${EXTEN}
with ${~~EXTEN~~} instead in application arguments; (and the ${EXTEN: also).
The reason for this, is that because switches are coded using
separate extensions to provide pattern matching, and
jumping to/from these switch extensions messes up the ${EXTEN} value,
which blows the minds of users.


........

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r111410 | murf | 2008-03-27 07:29:41 -0600 (Thu, 27 Mar 2008) | 17 lines

Merged revisions 111391 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r111391 | murf | 2008-03-27 07:03:28 -0600 (Thu, 27 Mar 2008) | 9 lines

These small documentation updates made in response to a query in
asterisk-users, where a user was using Playback, but needed the
features of Background, and had no idea that Background existed,
or that it might provide the features he needed. I thought the
best way to avert these kinds of queries was to provide "See Also"
references in all three of "Background", "Playback", "WaitExten".
Perhaps a project to do this with all related apps is in order.


........

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http://svn.digium.com/view/asterisk?view=rev&revision=111440