Joseph Shi [joseph@linksoft.com.hk] wrote:> (Article auto-converted from unnecessary HTML to nice plain text.)
>
> Does anybody know if the voice actually gets routed through Asterisk for
> calls between SIP devices? I just wonder if calls between SIP devices
> would take up any bandwidth or CPU at the Asterisk server. Please
> advise.
>
SIP devices will send re-invitations in an effort to find the most
efficient route for the voice data, bypassing the server(s) etc. In
a lot of cases, the two endpoints will end up speaking to one another
directly.
You can set up Asterisk to keep itself in the loop (canreinvite = no),
or it might want to remain in the loop regardless of your settings.
For instance, Asterisk will want to remain in the loop if you're
recording the call - for obvious reasons.
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