[asterisk-users] Re: SIP RTP flow

[Benny Amorsen]

>>>>> "MW" == Mike Williams <mike.williams at 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