[asterisk-bugs] [Asterisk 0010590]: [patch] RTP statistics returned by ast_rtp_get_quality reflects the last RTCP packet not a call overall

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=10590 
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Reported By:                gasparz
Assigned To:                bbryant
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Project:                    Asterisk
Issue ID:                   10590
Category:                   Core/NewFeature
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     closed
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
Resolution:                 fixed
Fixed in Version:           
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Date Submitted:             08-29-2007 06:22 CDT
Last Modified:              06-05-2008 11:19 CDT
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Summary:                    [patch] RTP statistics returned by
ast_rtp_get_quality reflects the last RTCP packet not a call overall
Description: 
The result of the function ast_rtp_get_quality is the value set by the LAST
RTCP frame. This value is inserted to the RTPAUDIOQOS channel variable and
should be a some kind of quality metric of the call.
So if we take only one sample: one (the last) RTCP frame we won't get a
relevant value for a parameter. The min/max values calculated (ex:
maxrtt,minrtt and all the others) are unused and would be very usefull to
aproximate call quality.
An average value could be easily computed for the parameters (ex: rtt) and
would give a mutch better picture of RTP quality.

Ex: 
Last rtt value: 80ms
Min rtt value: 30ms
Max rtt value: 500ms (perhaps congestion)
Avg rtt value: 35ms

My proposal is to create a new function:ast_rtp_get_quality_ext that would
return a detailed parsable string with the 4 values for each parameter.
This could be saved in another channel variable RTPAUDIOQOSEXT. This way we
would be backwards compatible with the current version.

Another less elegant way is to modify the curent function and add the
informations (min/max/avg) for each parameter at the end of the string
returned by ast_rtp_get_quality.




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---------------------------------------------------------------------- 
 svnbot - 06-05-08 11:19  
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Repository: asterisk
Revision: 120643

_U  branches/1.6.0/

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r120643 | bbryant | 2008-06-05 11:19:22 -0500 (Thu, 05 Jun 2008) | 21
lines

Blocked revisions 120635 via svnmerge

........
r120635 | bbryant | 2008-06-05 11:24:19 -0500 (Thu, 05 Jun 2008) | 14
lines

This patch adds more detailed statistics for RTP channels, and provides an
API call to access it, including maximums, minimums, standard deviatinos, 
and normal deviations. Currently this is implemented for chan_sip, but
could be added to the func_channel_read callbacks for the CHANNEL function 
for any channel that uses RTP.

(closes issue http://bugs.digium.com/view.php?id=10590)
Reported by: gasparz
Patches:
      chan_sip_c.diff uploaded by gasparz (license 219)
      rtp_c.diff uploaded by gasparz (license 219)
      rtp_h.diff uploaded by gasparz (license 219)
      audioqos-trunk.diff uploaded by snuffy (license 35)
      rtpqos-trunk-r119891.diff uploaded by sergee (license 138)
Tested by: jsmith, gasparz, snuffy, marsosa, chappell, sergee

........

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http://svn.digium.com/view/asterisk?view=rev&revision=120643 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
06-05-08 11:19  svnbot         Checkin                                      
06-05-08 11:19  svnbot         Note Added: 0087853                          
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