Steve Kann wrote:
Paul Fielding wrote:
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).
I've also got a Vonage line, using a Linksys ATA.
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.
Meanwhile the Vonage ATA maintains an almost flawless connection, all the time.
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?
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 (?).
A better jitterbuffer and Packet Loss Concealment is what you need.
It's coming to asterisk soon.
http://bugs.digium.com/bug_view_page.php?bug_id=0002532
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I would kill for this to be implemented! It's sorely needed for us folks that use Transatlantic lines etc. /Danny _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
