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.