<div>For the case where Asterisk is handling RTP bridging, and not TDM/DAC conversion, its not a jitter buffer you need but real-time platform performance.</div>
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<div>If Asterisk is the cause of jitter when its relaying media then its a flaw in Asterisk, just like a slow/broken router will cause jitter when forwarding from one interface to another.</div>
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<div>The fix is not to add a jitter buffer, but quite the opposite and all buffering and delays should be minimized.</div>
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<div>You would need to insure that real-time priority for the entire RTP path though Asterisk is guaranteed. I expect potential deadlock conditions would require much work for this to be successful.</div>
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<div>Mark <br><br> </div>
<div><span class="gmail_quote">On 10/31/05, <b class="gmail_sendername">Juliano Duque da Silva</b> <<a href="mailto:juliano.duque@terra.com.br">juliano.duque@terra.com.br</a>> wrote:</span>
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<p><font face="Arial" color="navy" size="2"><span lang="EN-US" style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">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.
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<p><font face="Arial" color="navy" size="2"><span lang="EN-US" style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">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" ?
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<p><font face="Arial" color="navy" size="2"><span lang="EN-US" style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Juliano </span></font></p>
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<p><b><font face="Tahoma" size="2"><span lang="EN-US" style="FONT-WEIGHT: bold; FONT-SIZE: 10pt; FONT-FAMILY: Tahoma">De:</span></font></b><font face="Tahoma" size="2"><span lang="EN-US" style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma">
<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:serusers-bounces@iptel.org" target="_blank">serusers-bounces@lists.iptel.org</a> [mailto:<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:serusers-bounces@lists.iptel.org" target="_blank">
serusers-bounces@lists.iptel.org</a>] <b><span style="FONT-WEIGHT: bold">Em nome de </span></b>Mark Aiken<br><b><span style="FONT-WEIGHT: bold">Enviada em:</span></b> domingo, 30 de outubro </span></font><font face="Tahoma" size="2">
<span style="FONT-SIZE: 10pt; 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> <a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:serusers@lists.iptel.org" target="_blank">
serusers@lists.iptel.org</a><br><b><span style="FONT-WEIGHT: bold">Assunto:</span></b> Re: [Serusers] Inserting SER into my voice network</span></font></p></div>
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<p style="MARGIN-BOTTOM: 12pt"><font face="Times New Roman" size="3"><span style="FONT-SIZE: 12pt">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.
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<p><span><font face="Times New Roman" size="3"><span style="FONT-SIZE: 12pt">On 10/30/05, <b><span style="FONT-WEIGHT: bold">Ray Van Dolson</span></b> <<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:rayvd@digitalpath.net" target="_blank">
rayvd@digitalpath.net</a>> wrote:</span></font></span> </p>
<p><font face="Times New Roman" size="3"><span style="FONT-SIZE: 12pt">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>_______________________________________________<br>Serusers mailing list<br><a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:Serusers@iptel.org" target="_blank">
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