asterisk users - Jun 2012 - Clipping issue with SIP over satellite

I'm having a wierd clipping issue with one employee who's using a phone
over a satellite Internet.  He was sold that system specifically for use
with VoIP.  Ping times show average round-trip time as around 700 ms with a
range of 560 to 841, so considerable jitter.

Things work fine when he's talking to another Asterisk phone or to a SIP
trunk provider, but when connecting to a T1, there's clipping where about
1/3 of his voice (in intervals of maybe 200ms) are removed.  This sounds
like an echo canceller conflict, but I've set echocancel=no in
chan_dahdi.conf (I have hardware echo cancelling) and it didn't do
anything.  I'm forcing his codec to G729 for bandwidth reasons.  The
phone is an Aastra 6757iCT.

Does anybody have any suggestions here?

On 06/17/2012 06:43 AM, Richard Kenner wrote:
> Things work fine when he's talking to another Asterisk phone or to a
SIP
> trunk provider, but when connecting to a T1, there's clipping where
about
> 1/3 of his voice (in intervals of maybe 200ms) are removed.  This sounds
> like an echo canceller conflict, but I've set echocancel=no in
> chan_dahdi.conf (I have hardware echo cancelling) and it didn't do
> anything.  I'm forcing his codec to G729 for bandwidth reasons.  The
> phone is an Aastra 6757iCT.
You have hardware echo canceling *outside* of your T1 card? If it's an 
echo canceler on the card, then setting 'echocancel=no' disables it. You
probably don't want to do that.

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.

As long as what are dealing with is 'simple' jitter (just delayed 
packets), as opposed to packet reordering, then this should help quite a 
bit. If you have packet reordering occurring as well, then you'll need a 
full-fledged adaptive jitter buffer on the channel to compensate for it. 
In recent releases of Asterisk, this can be done by using the 
JITTERBUFFER() dialplan function on the SIP channel in question, but 
since you didn't mention your version of Asterisk, I can't speculate 
whether that is available to you or not.
> Does anybody have any suggestions here?
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.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
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