There are 2 issues here.
1) Asterisk does not have a RTP Jitter Buffer. RTP is what is used to
transport audio for SIP (and other protocols). This means that ANY
jitter on the SIP Phone -> Asterisk link will cause audio problems.
2) Asterisk times it's outgoing audio based on the incoming audio.
Therefore, if there is jitter on the SIP Phone -> Asterisk link then
Asterisk will replicate that jitter on the Asterisk -> SIP Phone direction.
REMEMBER, a jitter buffer only applies on INCOMING audio (from the
standpoint of the device).
These two issues are the main reason I have not deployed remote SIP
phones for my clients.
So, he should probably try an IAX softphone and see how that compare
hth
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