[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:                     assigned
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             08-29-2007 06:22 CDT
Last Modified:              06-03-2008 03:33 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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---------------------------------------------------------------------- 
 sergee - 06-03-08 03:33  
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This patch has a bug, it would crash your asterisk if one of your call-legs
doesn't use RTP.

Backtrace:
http://bugs.digium.com/view.php?id=0  handle_request_bye (p=0x8205eb8,
req=0x82061c4) at chan_sip.c:14904
http://bugs.digium.com/view.php?id=1  0xb670756e in handle_request (p=0x8205eb8,
req=0xb66acfbc,
sin=0xb66ae2e0, recount=0xb66ae2f0, nounlock=0xb66ae2f4) at
chan_sip.c:15498
http://bugs.digium.com/view.php?id=2  0xb67094da in sipsock_read (id=0x81b6cd0,
fd=24, events=1, ignore=0x0)
at chan_sip.c:15638
http://bugs.digium.com/view.php?id=3  0x080ab610 in ast_io_wait (ioc=0x81b38e0,
howlong=1000) at io.c:279
http://bugs.digium.com/view.php?id=4  0xb66f07f8 in do_monitor (data=0x0) at
chan_sip.c:15851
http://bugs.digium.com/view.php?id=5  0x080fc57b in dummy_start (data=0x81b69f0)
at utils.c:895
http://bugs.digium.com/view.php?id=6  0xb7f0834b in start_thread () from
/lib/libpthread.so.0
http://bugs.digium.com/view.php?id=7  0xb71e465e in clone () from /lib/libc.so.6


Dialplan:

exten => 555,1,Answer();
exten => 555,n,MusicOnHold()
exten => 444,1,DIAL(IAX2/127.0.0.1/555)


SIP phone calls 444 - asterisk segfaults on hangup. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
06-03-08 03:33  sergee         Note Added: 0087700                          
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