On Apr 30, 2006, at 9:03 AM, 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.
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.
In my experience, even if you have two asterisk systems with the
async timing patch applied and are using IAX with the jitter buffer
enabled, asterisk STILL cannot compensate for jittery links as well
as Skype can. I take this to mean that the asterisk jitter buffer
needs more work.
In addition to having a better jitter buffer, Skype also clearly has
wideband codecs which sounds better.
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