We count a RTT at each point where the sender has to wait for a response before it can proceed. If desired, SIP can run directly on top of any protocol offering reliable or unreliable byte stream or datagram services, including ATM AAL5, IPX, X.25, without changes to the protocol. H.323 requires the use of a reliable transport protocol. Address-Format Neutrality SIP destination and forwarding addresses can be any URL, including mailto, phone, H.323 and http URL, affording flexibility in combining SIP with other signaling protocols. Email-like names can be mapped by any device on the Internet.
All aliases have to be resolved through a gatekeeper.
Since it uses UDP rather than TCP, SIP requests can be sent via multicast. H.323 has a separate spec, H.332, for including multicast listeners. The MC then decides whether to use multicast, and informs the endpoints of the addresses to use. As to invitation, normally a member of the conference asks the MC to invite somebody else. The Via header allows arbitrary routing of requests with loop detection.