I think you guys might have missed the original point. The answer to the question is, there is no SIP central registry because SIP is not an address resolution protocol. SIP is only a gateway protocol that allows a client at any IP address to connect with a given server, which then can connect with the PSTN. All the protocol does is to inform the server that a client is authenticated as having a certain IP address, user name and set of capabilities. The server then acts as that client's proxy to originate and terminate calls on the PSTN.
Now what you might be referring to is the handoff, if one SIP phone calls another. According to its dialplan, the SIP server knows that a certain dialed number can be reached at a certain SIP address (which is a regular IPv4 internet address) and extension. Once the calling and called servers agree, the SIP phones are told to make their own native bridge and the server drops out of the voice path.
As other posters have said, this is in its infancy and the applications are not mature. SIP is definitely a standard, though, and can be read at
https://www.ietf.org/rfc/rfc3261.txt?number=3261
