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Paul Fielding wrote:
<blockquote cite="mid01f601c4fd23$3b570d60$0400a8c0@MATHILDA"
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<div><font face="Arial" size="2">So far in my playing with Asterisk
I've messed with soft phones (x-ten, sjphone), hard phones (Grandstream
102), and ATA adapters (Grandstream 286, Digium IAXy).</font></div>
<div> </div>
<div><font face="Arial" size="2">I've also got a Vonage line, using a
Linksys ATA.</font></div>
<div> </div>
<div><font face="Arial" size="2">None of the devices I've connected
to my Asterisk server have been able to maintain the same consistent
sound quality over a long distance as the Vonage line. Don't get me
wrong, the Grandstreams are actually not too bad, but there is still
some breakups that can be annoying.</font></div>
<div> </div>
<div><font face="Arial" size="2">Meanwhile the Vonage ATA maintains
an almost flawless connection, all the time.</font></div>
<div> </div>
<div><font face="Arial" size="2">I'm assuming (perhaps wrongly?) that
the Linksys ATA that Vonage uses is still using SIP with some
standardized codec. If that assumption is correct, then how the heck
to they manage to get the consistent connection quality? Is it just a
matter of the right setting tweaks within Asterisk and/or the SIP
devices?</font></div>
<div> </div>
<div><font face="Arial" size="2">I don't think it's a question of
Asterisk hardware, since if I connect via local network to the Asterisk
server with a SIP device the quality is pretty consistent. It's
generally when remotely connecting that I have the inconsistent sound
quality. This would lead me to believe that it's a matter of tweaking
something to deal with latency or packet dropping issues (?).</font></div>
</blockquote>
A better jitterbuffer and Packet Loss Concealment is what you need.<br>
<br>
It's coming to asterisk soon.<br>
<br>
<a class="moz-txt-link-freetext" href="http://bugs.digium.com/bug_view_page.php?bug_id=0002532">http://bugs.digium.com/bug_view_page.php?bug_id=0002532</a><br>
<br>
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