[asterisk-users] Clipping issue with SIP over satellite

[Richard Kenner]

> You have hardware echo canceling *outside* of your T1 card? 

No, on the card.

> The DAHDI layer has some buffering that can help with jitter, but the 
> default buffers can only handle 80ms of jitter. You can increase this by 
> setting the 'buffers' option in chan_dahdi.conf; each buffer is 20ms by 
> default.

I'm running 1.6.2 and it appears that this is called jitterbuffers there.
Is that right?

I've set it to 20 and it did indeed help quite a bit, so I tried 30.

> It sounds like the lack of a proper jitter buffer (of adequate size) is 
> the issue here, since when the audio is directed at endpoints outside of 
> Asterisk that have them, the audio is as you'd expect it to be.

Interestingly, that isn't completely true.  If it goes out a SIP trunk
to PSTN, it works fine, but when it goes out a SIP trunk to the SV8300
(where the T1 goes), it has the same problem.  This was leading me to
believe that the problem was on the 8300.