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<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-US
style='font-size:10.0pt;font-family:Arial;color:navy'>I have a similar call
quality problem that Ray related when using asterisk is in the middle of media
path specially, when asterisk is performing codec transcoding. I am not sure
this problem is related to asterisk lack of jitter buffer but it seems to be.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-US
style='font-size:10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-US
style='font-size:10.0pt;font-family:Arial;color:navy'>How can we make sure
asterisk is not screwing up the jitter buffer? Someone on this list knows
exactly how asterisk performs the “RTP proxing” ? <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-US
style='font-size:10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-US
style='font-size:10.0pt;font-family:Arial;color:navy'>Juliano <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-US
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 lang=EN-US
style='font-size:10.0pt;font-family:Tahoma;font-weight:bold'>De:</span></font></b><font
size=2 face=Tahoma><span lang=EN-US style='font-size:10.0pt;font-family:Tahoma'>
serusers-bounces@iptel.org [mailto:serusers-bounces@lists.iptel.org] <st1:PersonName
ProductID="Em nome de Mark Aiken" w:st="on"><b><span style='font-weight:bold'>Em
nome de </span></b>Mark Aiken</st1:PersonName><br>
<b><span style='font-weight:bold'>Enviada em:</span></b> domingo, 30 de outubro
</span></font><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma'>de 2005 17:41<br>
<b><span style='font-weight:bold'>Para:</span></b> Ray Van Dolson<br>
<b><span style='font-weight:bold'>Cc:</span></b> serusers@lists.iptel.org<br>
<b><span style='font-weight:bold'>Assunto:</span></b> Re: [Serusers] Inserting
SER into my voice network</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 style='margin-bottom:12.0pt'><font size=3
face="Times New Roman"><span style='font-size:12.0pt'>I cant see that at all
from your diagram. I see only an ATA and Media Gateway doing final conversion
where jitter buffer would be useful. If turing on a jitter buffer in Asterisk
helps then one of the other 2 is broke.<o:p></o:p></span></font></p>
<div>
<p class=MsoNormal><span class=gmailquote><font size=3 face="Times New Roman"><span
style='font-size:12.0pt'>On 10/30/05, <b><span style='font-weight:bold'>Ray Van
Dolson</span></b> <<a href="mailto:rayvd@digitalpath.net">rayvd@digitalpath.net</a>>
wrote:</span></font></span> <o:p></o:p></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>When I take Asterisk out of the media path, this is
correct. And I believe my<br>
ISP's media gateway *does* have a jitter buffer. <br>
<br>
Since Asterisk was an media endpoint before (it doesn't just proxy the rtp<br>
on), its lack of jitter buffer was hurting us in some cases.<br>
<br>
Ray<br>
<br>
On Sun, Oct 30, 2005 at 08:55:13AM -0600, Mark Aiken wrote: <br>
><br>
> The only jitter buffers that matter in your diagram
are the SIP ATA and<br>
> Media Gateway. Both should have jitter buffers at
the point where they<br>
> convert RTP to PCM. If adding a jitter
buffer inside the network path <br>
> somewhere helps then something else is broken.<br>
><br>
><br>
><br>
> Mark<br>
<br>
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<p class=MsoNormal style='margin-bottom:12.0pt'><font size=3
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