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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Hi,<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'> Forgot to mention that I am using
Asterisk version 1.4.15 running on RHEL 3.0 server.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Regards,<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Mayur<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'> Mayur
[mailto:mninama@varaha.com] <br>
<b><span style='font-weight:bold'>Sent:</span></b> Wednesday, December 26, 2007
7:03 PM<br>
<b><span style='font-weight:bold'>To:</span></b> '<st1:PersonName w:st="on">asterisk-users@lists.digium.com</st1:PersonName>'<br>
<b><span style='font-weight:bold'>Subject:</span></b> SIP Channel jitter buffer
issue</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Hi,<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> I have a SIP client which is registered to
asterisk. Asterisk is registered to a SIP trunk and also handles the media. Now
since my client has some issues in its RTP Tx, which seems to have some amount
of jitter (mean jitter as per ethereal trace is about 17ms, max jitter is 20 ms
and max delta is 85 ms), to over come that I have enabled jitter buffer in the
SIP channel by setting sip.conf parameters jenable=yes, jbforce=yes,
jbmaxsize=200 and jbimpl=fixed. However on setting these parameters I am unable
to hear on the trunk side. From the jitter logs as given below, I can see audio
frames being dropped:<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>JB_PUT {now=1130}: Dropped frame with ts=21125 and len=20<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>
JB_GET {now=1130}: now < next=2121<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>
JB_GET {now=1142}: now < next=2121<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>
JB_GET {now=1163}: now < next=2121<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>JB_PUT {now=1181}: Dropped frame with ts=21132 and len=20<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>
JB_GET {now=1181}: now < next=2121<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>
JB_GET {now=1183}: now < next=2121<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>JB_PUT {now=1185}: Dropped frame with ts=21132 and len=20<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>
JB_GET {now=1185}: now < next=2121 <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>I have tried increasing the jitter buffer from 200 to 1000
ms but with same result. <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Am I missing anything here? How can I determine what is
causing asterisk to drop the audio frames?<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Regards,<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Mayur<o:p></o:p></span></font></p>
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