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

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=11035 
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Reported By:                praeter
Assigned To:                
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Project:                    Asterisk
Issue ID:                   11035
Category:                   Applications/app_record
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
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:              
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Date Submitted:             10-19-2007 06:50 CDT
Last Modified:              11-06-2007 14:06 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. 
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---------------------------------------------------------------------- 
 praeter - 11-06-07 14:06  
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It is all MoH. Three seconds elapsed actual time in MoH results in 3:44 of
recorded MoH. As I have said, the data rate changes as soon as it is an
asterisk to asterisk communication. As soon as an endpoint(telephone) is
reinserted in the process the data rate throttles back to 64K. There has
got to be a better way to communicate with you. A telephone call...or
something. This communication is not working because I can't seem to get
across to you as hard as I try. 

Have you plotted the attached file? Is there a way I can send you a copy
of the recorded file? Is there someone else who can review the issue? Have
you attempted to replicate the issue? I can do it every time. Help me help
you. 

Issue History 
Date Modified   Username       Field                    Change               
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11-06-07 14:06  praeter        Note Added: 0073223                          
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