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 (MCUs), 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 (URLs) can also
be embedded in web pages. SIP is addressing-neutral, with addresses
expressed as URLs 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.
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