I did find the jitter buffer patch on the bug-tracker...(ast_jb-1.2.0.patch4). I applied it to a 1.2.6 asterisk and it seemed to apply all but 2 small chunks (which I was able to apply myself)... it then compiled... so I'm going to give it a shot and test it out. I will report back results.
<br><br><div><span class="gmail_quote">On 4/11/07, <b class="gmail_sendername">Alex Balashov</b> <<a href="mailto:abalashov@evaristesys.com">abalashov@evaristesys.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
On Wed, 11 Apr 2007, Matt said something to this effect:<br><br>> I know that there was a jitter buffer patch (for sip) for the 1.0.9 branch<br>> some time agin. At this time, we can not upgrade to 1.4.x. Is there a
<br>> useable, fairly stable INCOMING sip jitter buffer patch? That is.. I want<br>> Asterisk to jitter buffer incoming SIP packets.<br><br> Incoming RTP packets you mean? :-)<br><br> I am not aware that a jitter buffer patch exists for
1.0.x. But I could<br>be wrong; however, when I ran into this exact same issue I was not able to<br>find anything. But I didn't try very hard.<br><br> This is a question you may possibly want to ask on the asterisk-dev list.
<br><br>-- Alex<br><br>--<br>Alex Balashov <<a href="mailto:sasha@presidium.org">sasha@presidium.org</a>><br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">
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