[Asterisk-Users] Compare to Skype

[Time Bandit]

> 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.

So, he should probably try an IAX softphone and see how that compare

hth