[Asterisk-Users] Compare to Skype

[Richard Scobie]

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