[asterisk-bugs] [Asterisk 0015819]: [patch] buggy output in "sip show channelstats"

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=15819 
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Reported By:                klaus3000
Assigned To:                oej
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Project:                    Asterisk
Issue ID:                   15819
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     closed
Asterisk Version:           1.6.2.0-beta4 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 fixed
Fixed in Version:           
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Date Submitted:             2009-09-02 12:09 CDT
Last Modified:              2010-01-13 05:25 CST
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Summary:                    [patch] buggy output in "sip show channelstats"
Description: 
Hi!

"sip show channelstats" output is wrong
- the packetloss in % is always 0 due to integer division instead of float
divison
- the local measured jitter is reported in as rxjitter (which in in
seconds) * 100, converted to int. Multiply with 100 makes no sense - I
supsect a type and it should be 1000 as all other jittervalues (max/min...)
are also multiplyed with 1000.
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Relationships       ID      Summary
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related to          0015860 wrong parsing of received RTCP packets
related to          0015807 rtt should be stored as double in struc...
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 (0116564) svnbot (reporter) - 2010-01-13 05:25
 https://issues.asterisk.org/view.php?id=15819#c116564 
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Repository: asterisk
Revision: 239703

U   branches/1.6.2/channels/chan_sip.c

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r239703 | oej | 2010-01-13 05:25:56 -0600 (Wed, 13 Jan 2010) | 14 lines

Show proper stats in "sip show channelstats"

SIP show channelstats show current RTCP statistics for calls - if we have
it. Calls bridged
in RTP p2p bridge doesn't have any statistics.  In calls where the remote
end doesn't send
RTCP or we can't receive it due to NAT, there's no reliable data as well.

Thanks, Klaus, for the patch. Sorry for the delay.
(closes issue https://issues.asterisk.org/view.php?id=15819)
Reported by: klaus3000
Patches: 
      asterisk-sip-show-channelstats-1.6.2.txt uploaded by klaus3000
(license 65)
Tested by: klaus3000, oej


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http://svn.digium.com/view/asterisk?view=rev&revision=239703 

Issue History 
Date Modified    Username       Field                    Change               
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2010-01-13 05:25 svnbot         Checkin                                      
2010-01-13 05:25 svnbot         Note Added: 0116564                          
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