[asterisk-bugs] [Asterisk 0011035]: Record on sip trunk does not maintian voice codec data rate

noreply at bugs.digium.com noreply at bugs.digium.com
Tue Nov 6 14:19:23 CST 2007


The following issue has been RESOLVED. 
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http://bugs.digium.com/view.php?id=11035 
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Reported By:                praeter
Assigned To:                qwell
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Project:                    Asterisk
Issue ID:                   11035
Category:                   Applications/app_record
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     resolved
Asterisk Version:            1.4.10.1  
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
Resolution:                 won't fix
Fixed in Version:           
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Date Submitted:             10-19-2007 06:50 CDT
Last Modified:              11-06-2007 14:19 CST
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Summary:                    Record on sip trunk does not maintian voice codec
data rate
Description: 
Scenario: Two asterisk systems; one is sip source(asterisk 1.2.16) the
other is unit under test(asterisk 1.4.10.1). Unit under test is setup to
answer all incoming calls and immediately begins recording for the duration
of the SIP session. If a call is placed to the unit under test, spoken
voice is first sent (test 1 2 3), then the call is placed on hold for three
seconds, and then a final  spoken voice (end 1 2 3) for a total call
duration of 19 seconds (verified by cdr record). The subsequent  recording
is 4 minutes(verified by listening to entire recording; no skips or
silence). If a call is placed to the unit under test and only spoken voice
is done, recording = call duration. This is telling me that the data rate
for pure system to system audio is different than live voice. The same
condition exists if the milliwatt signal is sent instead of MoH. Recorded
file exceeds call duration by x5. 
====================================================================== 

---------------------------------------------------------------------- 
 qwell - 11-06-07 14:19  
---------------------------------------------------------------------- 
This is an issue with Asterisk 1.2.  Please try 1.4 (on both sides), and
reopen if you can still reproduce. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
11-06-07 14:19  qwell          Status                   new => resolved     
11-06-07 14:19  qwell          Resolution               open => won't fix   
11-06-07 14:19  qwell          Assigned To               => qwell           
11-06-07 14:19  qwell          Note Added: 0073224                          
======================================================================




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