[asterisk-bugs] [Asterisk 0016663]: RTP Timeout is flawed
Asterisk Bug Tracker
noreply at bugs.digium.com
Thu Jan 21 07:37:41 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 07:37 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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(0117010) falves11 (reporter) - 2010-01-21 07:37
https://issues.asterisk.org/view.php?id=16663#c117010
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Could you supply a session-times configuration for sip.conf? This is what I
do: I get a SIP call and place an outbound call also via SIP. I need to
send packets to the caller and to the callee to prevent that either is
dead, otherwise the billing is useleess, basically. I was using rtptimeout
but as you can see, under some pretty normal conditions, it is quite
useless. I use 1.4. I wonder if it is possible indeed to achieve what I
need to achieve with 1.4.
Issue History
Date Modified Username Field Change
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2010-01-21 07:37 falves11 Note Added: 0117010
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