[asterisk-users] Clipping issue with SIP over satellite

[Valer Nur]

>> 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.

You can try and improve audio quality and "compensate" on the problems with the 'SV8300' device by using quality improvement software like PBXMate. It should be able and take care of server-side echo cancellation.
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