[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:                     acknowledged
Asterisk Version:           1.6.2.0-beta4 
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:             2009-09-02 12:09 CDT
Last Modified:              2009-09-05 01:27 CDT
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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          0015807 rtt should be stored as double in struc...
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 (0110248) oej (manager) - 2009-09-05 01:27
 https://issues.asterisk.org/view.php?id=15819#c110248 
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The p2p bridge is an optimized bridge that works like the good ol' RTPproxy
more or less. I don't know all the details, I think it terminates RTP
streams, and forwards on RTP layer more than UDP layer, but never parses
the content of either RTP or RTCP. It for situations where we can't use the
remote RTP functionality (directmedia/reinvite stuff) because of NAT, but
have no requirements to actually be involved in the content (no recordings
or such).

Writing this I don't know whether that bridge can still parse DTMF. But
that's a separate issue.

Yes, the memset sounds fine, I never checked code on how complicated
"initializing rtcp" was, but if a memset is all it takes, then we'll move
it. 

It's always easier to do things in a series of small patches, so I can
merge the memset a.s.a.p so we're done with that. As soon as we have a
patch. I see it as a bug, so we can merge it from 1.4 and up. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-09-05 01:27 oej            Note Added: 0110248                          
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