[asterisk-users] SIP RTP flow

[Moises Silva]

You can make RTP pass through Asterisk, or not. Look in voip-info.org
about "Native Bridge" and "sip.conf" "canreinvite" option. And may be
this page will be usefull too:

http://www.voip-info.org/wiki/index.php?page=Asterisk+SIP+not-proxy

Regards

On 10/31/06, Mike Williams <mike.williams at comodo.com> wrote:
> Hey,
>
> This is probably a rather stilly question...
>
> If I pick up my SIP phone that's registered to my asterisk server and dial a
> number that asterisk recognises as destined for a SIP trunk (could be a
> static route, or an ENUM lookup) or another SIP device registered on said
> asterisk server (internal extension to extension call), what route does the
> actual audio take?
>
> The control connection (port 5060) obviously goes via the asterisk server as
> it has to work out where to send the control to, but I could quite easily
> imagine the audio going directly handset to remote server or handset to
> asterisk to remote, and handset to handset or handset to asterisk to handset.
>
> Thanks
>
> --
> Mike Williams
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