A discussion on A SIP call Sequence

We were having a discussion on how a SIP call is processed through an Asterisk system. Lets say we have a system on a network and a phone connected to it. Also a remote phone, connected to the system and several SIP trunks. When the Remote user needs to dial out on the SIP trunks they obviously will use a trunk on the system. After the initial call setup by the system, does the Asterisk system hand off ALL the call bandwidth between the phone and the sip trunk provider. Or does the System act as a “Bridge” between the two?

Is there a resource that will explain this??

Thanks alot

Thanks Tony for the reply.

By default it keeps the media. Part of being a B2BUA. You can turn off the feature so it does not proxy the media but you will loose features then.