[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
Wed Dec 3 18:51:52 CST 2008
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=13801
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Reported By: justdave
Assigned To: Corydon76
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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:
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Date Submitted: 2008-10-29 13:47 CDT
Last Modified: 2008-12-03 18:51 CST
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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.
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(0095763) DEA (reporter) - 2008-12-03 18:51
http://bugs.digium.com/view.php?id=13801#c95763
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mdu113/justdave, can either of you share some details about your
environment? I found a interesting bit while working on this
today.
During an RTP debug I spotted the fact (if the ts value can be
trusted) that we were sending 30ms of audio every 40ms. The 30ms
was desired, but not at a 40ms interval. I reconfigured my PSTN
gateways back to 20ms, and the quality improved dramatically.
Issue History
Date Modified Username Field Change
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2008-12-03 18:51 DEA Note Added: 0095763
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