[asterisk-users] Clipping issue with SIP over satellite

[Kevin P. Fleming]

On 06/19/2012 04:23 AM, Richard Kenner wrote:
>> You have hardware echo canceling *outside* of your T1 card?
>
> No, on the card.

Then you definitely don't want 'echocancel=no' set, or you'll disable it.

>
>> 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?

Yes.

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

Excellent!

>
>> 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.

Well, that doesn't disprove my statement :-) Note that I said that when 
the audio is directed at endpoints that have a proper jitter buffer, 
there is no issue. If you send the call over SIP to this 'SV8300' device 
and still have audio issues, that would imply that this device does not 
have a jitter buffer capable of handling this level of jitter.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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