[asterisk-bugs] [Asterisk 0014295]: SIP on hold problems

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=14295 
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Reported By:                klaus3000
Assigned To:                file
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Project:                    Asterisk
Issue ID:                   14295
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     assigned
Asterisk Version:           1.4.22 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-01-21 07:20 CST
Last Modified:              2009-01-22 13:09 CST
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Summary:                    SIP on hold problems
Description: 
Hi!

I have found 2 problems when the SIP phone puts a call on hold (a SNOM
phone which uses a=sendonly).

1. Although only RTP should be activated it looks like RTCP is deactivated
too, as I got these error message on the console:

  RTCP SR transmission error, rtcp halted

2. When the call is on hold and I unplug the phone, I would think that the
session is terminated after "rtpholdtimeout". Instead the session is
termianted after rtptimeout.

see trace below: rtptimeout=15, rtpholdtimeout=25
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Relationships       ID      Summary
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related to          0013835 "RTCP SR transmission error, rtcp ...
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 (0098471) file (administrator) - 2009-01-22 13:09
 http://bugs.digium.com/view.php?id=14295#c98471 
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The second issue about the RTCP log message has been solved already, I am
investigating the first. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-01-22 13:09 file           Note Added: 0098471                          
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