[asterisk-bugs] [Asterisk 0013801]: [patch] No way to tune talker optimization in meetme, causes users to get cut off while they're still talking

Asterisk Bug Tracker noreply at bugs.digium.com
Tue Nov 25 14:10:59 CST 2008


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=13801 
====================================================================== 
Reported By:                justdave
Assigned To:                Corydon76
====================================================================== 
Project:                    Asterisk
Issue ID:                   13801
Category:                   Applications/app_meetme
Reproducibility:            have not tried
Severity:                   major
Priority:                   normal
Status:                     confirmed
Asterisk Version:           1.4.22 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             2008-10-29 13:47 CDT
Last Modified:              2008-11-25 14:10 CST
====================================================================== 
Summary:                    [patch] No way to tune talker optimization in
meetme, causes users to get cut off while they're still talking
Description: 
I enabled 'o' talker optimization on my conference rooms because the
documentation in 1.4 says the feature will be permanently enabled in
Asterisk 1.6 with no way to disable it, so I figured we should probably get
used to it. However, if it works like this we'll have to never upgrade to
1.6.  We get constant complaints about people getting cut off while still
talking in the conferences, and I can't find any way to tune what it
considers "talking". If the feature is going to be permanently enabled, we
at least need some way to tune how sensitive it is.
====================================================================== 

---------------------------------------------------------------------- 
 (0095495) DEA (reporter) - 2008-11-25 14:10
 http://bugs.digium.com/view.php?id=13801#c95495 
---------------------------------------------------------------------- 
No real progress, but another set of observations that may help-

1.  The quality seems to fall apart after the 3rd caller joins.
The first two callers heard clear audio, everyone else hears
the warble.
2.  1.6 now seems to mux in comfort noise/background noise.  1.4
produces crystal clear silence, even from noisy cell phones.  Maybe
the bridge/DSP is working to hard on useless audio.
3.  ztdummy/zttest score higher on accuracy than dahdi_dummy/dahdi_test

My environment does not have hardware timing source, only *dummy
interfaces

I rolled the productio system back, but I have a test system that I can 
continue to test possible fixes on. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-11-25 14:10 DEA            Note Added: 0095495                          
======================================================================




More information about the asterisk-bugs mailing list