[asterisk-bugs] [Asterisk 0011562]: "rtptimeout" doesn't terminate channel if RTP is lost during "Echo()"

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=11562 
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Reported By:                ibc
Assigned To:                file
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Project:                    Asterisk
Issue ID:                   11562
Category:                   Channels/chan_sip/*
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     assigned
Asterisk Version:            SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 93152 
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             12-15-2007 08:58 CST
Last Modified:              01-05-2008 06:13 CST
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Summary:                    "rtptimeout" doesn't terminate channel if RTP is
lost during "Echo()"
Description: 
I set rtptimeout=20 in sip.conf and call from a softphone to extension
501:

  exten => 501,1,Answer
  exten => 501,n,Echo

During the call I kill the softphone (RTP ended and SIP BYE never received
by Asterisk).

After a while (qualify=yes) softphone peer appears as UNREACHABLE.
Asterisk gets "ICMP host UNREACHABLE" when sending RTP to client.
And 20 seconds after (rtptimeout) nothing occurs, the channel remains open
and will never be hangup. Why?
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---------------------------------------------------------------------- 
 rjain - 01-05-08 06:13  
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ibc, 

You may want to try SIP session-timers:
http://bugs.digium.com/view.php?id=10665

rtptimeout and rtpholdtimeout have certain limitations that do not exist
w/ SIP session-timers. 

Issue History 
Date Modified   Username       Field                    Change               
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01-05-08 06:13  rjain          Note Added: 0076340                          
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