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<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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