[Asterisk-Users] Compare to Skype

[Matt Ranney]

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.