[Asterisk-Users] SIP/iax routing question

[snacktime]

If I understand it correctly, SIP just handles the signalling between
endpoints.  When I call someone via a sip proxy, once the connection
is made all the audio is going directly from me to the person I am
calling correct?   What happens if a SIP call is routed through more
than one * server?


Also, when setting up an inter asterisk exchange, is all the data
routed through the * servers?

Chris