One of my user is praising Skype!!!
I cannot figure out anymore what I can improve!
This users sip show peers is jumping from 65 msec to 1800 all the
time. Of course his voice quality is like a morse code with dashes or
dots of connection time.
The next minute he calls me via Skype and it works fine !!!! What
indicates that there is no fault on his Internet connection!!!
He is using his notebook and Xlite, but also tried the snom 360.
Any hints?
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.
I believe that BOTH of these issues will be fixed in Asterisk 1.4.x,
which should be released sometime this summer.
--
Now accepting new clients in New Orleans, Birmingham, Atlanta,
Huntsville, Chattanooga, and Montgomery.
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