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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 (?).
What has Vonage got figured out that I still need
to? Any comments would be appreciated...
regards,
Paul
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