SIP is best described as a simple control protocol for creating, modifying and terminating sessions with one or more participants. Sessions could include internet multimedia conferences, internet telephone calls and full multimedia exchange. SIP supports session descriptions that will allow participants to agree on a set of preferred and compatible media formats. It will also support user mobility by proxying requests to the user’s current location. SIP itself is not associated to any conference control type protocols. In essence, SIP provides or enables the following functions:
Participant management of calls
During a call a participant can bring other users into the call or cancel connections to other users. In addition, users could be transferred or placed on hold etc.
Feature negotiation(s)
This allows the group(s) involved in a call to agree on the features to be supported (recognising that not all the parties can support the same level of features e.g. video may or may not be supported)
Changing call features
A user should be able to change the call characteristics during the actual course of the call. A call may have been set up as ‘voice-only’, but during the call, the users may need to enable video.
SIP borrows heavily from the email model, using the Domain Name System to deliver requests to the server that can handle them appropriately. This helps to simplify the integration of voice and email. Servers along the call path can then easily create and forward email messages, and vice versa, enabling true convergence.
SIP provides the protocol so that the end systems and proxies can provide services:
User location, user availability, call set-up, call handling, call forwarding no answer; call-forwarding on busy, call-forwarding un-conditional & other address-translation based services caller and calling “number” delivery (CLIP), where numbers can be any (preferably unique) naming scheme, personal mobility, i.e. the ability to reach a signal called party under a location-independent address even when the user changes terminals, terminal-type negotiation and selection i.e. a caller can now be given a choice to reach the party, e.g. via internet telephony, mobile phone, an answering service, etc.; terminal capability negotiation, caller and callee authentication and finally blind and supervised call transfer ability.
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