[asterisk-bugs] [Asterisk 0019399]: RTP Timestamp jump on marker packet

[Asterisk Bug Tracker]

The following issue has been SUBMITTED. 
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https://issues.asterisk.org/view.php?id=19399 
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Reported By:                iman
Assigned To:                
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Project:                    Asterisk
Issue ID:                   19399
Category:                   Core/RTP
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.8.4.1 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2011-06-01 20:30 CDT
Last Modified:              2011-06-01 20:30 CDT
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Summary:                    RTP Timestamp jump on marker packet
Description: 
I have been trying to track down some choppiness that I experience when my
server is connected to a SIP Trunk.  I set up a inbound call and directed
that to MilliWatt app to get continious tone.  When I call in, I hear the
tone with drop out every few seconds.  

I dig deeper and capture packet right in the server where the asterisk is
ran.  Please note that the packets captured do not leave physical network.

In that capture,  I found that on every RTP marker packet, the timestamp
jump more than I expected.  My understanding is that on every RTP packet
for G711u contain 20ms worth of audio.  The packet time stamp should
increase by rate of 160.  Please see the attached untitled.png.  You can
see that on that marker packet the timestamp jump. I use wireshark to open
the captured packet files (my.cap.gz).

I also use wireshark to analyze the packet.  See untitled2.png.  Note that
the decoded RTP packet has gap in it.  Note that in the Stream Analysis
screen that the skew value is very huge and the rate is 9201 HZ. ( I
thought that it should be like 8000HZ).  If you play the audio,  the sound
of the tone is close to what you hear by calling in to the sip trunk.

I have setup 2 DID SIP Trunk from two different company.  I can not make
that number public since it is a paid per minuite plan.  However,  I can
share that number with the developer working on this issue.  You can hear
that the tone drop out is similiar from the one that wireshark play out.

I also am attaching the asterisk log file during the call.  

Thanks
Iman

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Issue History 
Date Modified    Username       Field                    Change               
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2011-06-01 20:30 iman           New Issue                                    
2011-06-01 20:30 iman           Asterisk Version          => 1.8.4.1         
2011-06-01 20:30 iman           Regression                => No              
2011-06-01 20:30 iman           SVN Branch (only for SVN checkouts, not tarball
releases) => N/A             
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