asterisk users - Feb 2012 - SIP trunk audio bad but is OK again after SIP re-registration

Hi,

When *ANY* SIP client (softphone, hardphone, ATA) registers to an Asterisk
server on my LAN and the extension dials out through a remote SIP
provider, the audio is fine for "a while". It then degrades and starts
to be "cracky"/jittery. The extension can call once and again and it
will
always be bad. The only way to somehow "fix" the audio problem is to
unregister the local SIP extension/hardphone/softphone and register it back
to the same Asterisk server.

I repeated the test several times and it seems to be reproducible.

It apparently has nothing to do with my SIP provider or my DSL connection or
router. It doesn't even seem to be a network problem on my side.
Curiously though, it only happens if dialing out through the SIP provider...

I thought maybe the Asterisk server's system clock could be an issue but it
doesn't seem to be skewing off too quickly.

Also, this problem started showing up 2 weeks ago. Before that, we've been
making a lot of calls through the provider without a glitch. Nothing has changed
as far as hardware and software is concerned.

What could I try? How can I debug this?
Why is re-registering the SIP extension making a difference?
Any clues?

Asterisk 1.4

Thanks,

Vieri