asterisk users - Oct 2006 - SIP RTP flow

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

>>>>> "MW" == Mike Williams
<mike.williams@comodo.com> writes:
MW> The control connection (port 5060) obviously goes via the asterisk
MW> server as it has to work out where to send the control to, but I
MW> could quite easily imagine the audio going directly handset to
MW> remote server or handset to asterisk to remote, and handset to
MW> handset or handset to asterisk to handset.

Try looking for "reinvite" or "canreinvite", and you will be
enlightened.


/Benny

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@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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