[asterisk-bugs] [Asterisk 0016663]: RTP Timeout is flawed
Asterisk Bug Tracker
noreply at bugs.digium.com
Thu Jan 21 06:35:00 CST 2010
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=16663
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Reported By: falves11
Assigned To:
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Project: Asterisk
Issue ID: 16663
Category: Channels/chan_sip/Interoperability
Reproducibility: sometimes
Severity: major
Priority: normal
Status: new
Asterisk Version: 1.4.29
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): 1.4
SVN Revision (number only!): 238629
Request Review:
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Date Submitted: 2010-01-20 17:50 CST
Last Modified: 2010-01-21 06:35 CST
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Summary: RTP Timeout is flawed
Description:
The sip.conf configuration rtptimeout=XX only considers one way, not both.
Suppose the caller is silent for more than XX seconds, on mute, while his
mother issues a long speech. The call drops. This function must consider
audio both ways, and only if there is silence in both channels, drop the
call.
In my particular case, if I call to a voicemail where the greeting is
longer than XX seconds, and I simply keep my microphone silent, the call
will drop exactly at XX seconds, invalidating the whole thing.
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(0117008) falves11 (reporter) - 2010-01-21 06:35
https://issues.asterisk.org/view.php?id=16663#c117008
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rtpkeepalive actually breaks the compatibility with many switches. I cannot
use it. The calls actually fail in the first few seconds when talking to
many flavos of border controllers and softswitches.
The consequences of rtptimeout=XX are such that if I stop talking and just
listen, like in a conference where many people are in "mute", then the call
drops. This is irrational.
Issue History
Date Modified Username Field Change
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2010-01-21 06:35 falves11 Note Added: 0117008
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