[SIP] provides the necessary protocol mechanisms so that end systems and proxy servers can provide services: * call forwarding, including * the equivalent of 700-, 800- and 900- type calls; * call-forwarding no answer; * call-forwarding busy; * call-forwarding unconditional; * other address-translation services; * callee and calling ``number'' delivery, where numbers can be any (preferably unique) naming scheme; * personal mobility, i.e., the ability to reach a called party under a single, location-independent address even when the user changes terminals; * terminal-type negotiation and selection: a caller can be given a choice how to reach the party, e.g., via [Internet] telephony, mobile phone, an answering service, etc.; * terminal capability negotiation; * caller and callee authentication; * blind and supervised call transfer; * invitations to multicast conferences. Extensions of [SIP] to allow third-party signaling (e.g., for click-to-dial services, fully meshed conferences and connections to multipoint control units ([MCU]s), as well as mixed modes and the transition between those) are available. [SIP] addresses users by an email-like address and re-uses some of the infrastructure of electronic mail delivery such as [DNS] MX records or using [SMTP] [EXPN] for address expansion. [SIP] addresses ([URL]s) can also be embedded in web pages. [SIP] is addressing-neutral, with addresses expressed as [URL]s of various types such as [SIP], [H.323] or telephone ([E.164]). [SIP] can also be used for signaling [Internet] real-time fax delivery. This requires no major changes. Fax might be carried via [RTP], [TCP] (e.g., the protocols discussed in the [Internet] fax WG) or other mechanisms. [SIP] is independent of the packet layer and only requires an unreliable datagram service, as it provides its own reliability mechanism. While [SIP] typically is used over [UDP] or [TCP], it could, without technical changes, be run over [IPX], or carrier pigeons, frame relay, [ATM] [AAL5] or [X.25], in rough order of desireability.