Eric "ManxPower" Wieling wrote:
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.
This is only an issue if your SIP phone has a poor/nonexistent jitter
buffer.
The ideal scenario from a latency point of view is for the end points to
handle jitter buffering. I use Polycom 500's with G711 over a path where
jitter can be quite severe on occasion and they handle it very well.
Although I have not tried them, one would expect Cisco's to work well also.
Regards,
Richard
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