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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><a href="http://www.asterisk.org/node/48317">http://www.asterisk.org/node/48317</a>
does a nice job of explaining the 1.4 jitter buffer, however it raised a
question in my mind.<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'>In 1.2 (and also 1.4), when asterisk bridges 2 SIP channels,
are the UDP RTP packets renumbered on transmit, or is the original sequence
number preserved in the UDP header?<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'>A comment is made on the referenced blog that jitter
buffering is best implemented at the endpoint, where additional jitter will not
be added by another IP link. This is logical thinking, but only possible if the
bridging function in Asterisk preserves the source call leg UDP packet
numbering in the terminating call LEG UDP RTP packet stream.<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'>If the effect of the Asterisk SIP to SIP bridge is such that
the UDP headers are re-created on transmit it is likely that the packet
sequencing is the order in which Asterisk transmitted the packets, which is may
not be the order in which the original source UA transmitted them due to jitter
in the IP link on the first half of the bridged call.<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'>Can anyone provide an authoritative answer on how asterisk
sequences UDP RTP packets on the transmit leg of a bridged SIP call (known based
on actual testing or code review)? <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'>Or maybe there is information I lack that makes this a silly
question, such as where the SIP RTP sequence number is stored in the packet
(ie: not in the UDP header?) </span></font><font size=2 face=Wingdings><span
style='font-size:10.0pt;font-family:Wingdings'>J</span></font><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'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Thanks!<o:p></o:p></span></font></p>
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