SIP and Associated Standards
SIP is a signaling and control protocol for the establishment,
maintenance, and termination of multimedia sessions with one or more
participants. Examples of SIP multimedia sessions include Internet telephone
calls, multimedia conferences, and multimedia distribution. Session
communications might be based on multicast, unicast, or both.
SIP operates on the principle of session invitations. Through
invitations, SIP initiates sessions or invites participants into established
sessions. Descriptions of these sessions are advertised by any one of several
means, including the Session Announcement Protocol (SAP), defined in RFC 2974,
which incorporates a session description according to the Session Description
Protocol (SDP), defined in RFC 2327.
SIP uses other IETF protocols to define other aspects of VoIP
and multimedia sessions. For example, SIP can use URLs for addressing, DNS for
service location, and Telephony Routing over IP (TRIP) for call routing.
SIP supports personal mobility and
other intelligent ietwork (IN) telephony subscriber services through name
mapping and redirection services. Personal mobility allows a potential
participant in a session to be identified by a unique personal number or
name.
IN provides carriers with the ability to rapidly deploy new
user services on platforms that are external to the switching fabric. Access to
the external platforms is by way of an independent vendor and standard user
interface. Calling-card services, 1-800 services, and local number portability
are just three of these services.
Multimedia sessions are established and terminated by the
following services:
-
User location services Locate
an end system
-
User capabilities services
Select the media type and parameters
-
User availability services
Determine the availability and desire for a party to participate
-
Call setup services Establish
a session relationship between parties and manage call progress
-
Call handling services
Transfer and terminate calls
Although the IETF has made great progress in defining
extensions that allow SIP to work with legacy voice networks, the primary
motivation behind SIP is to create an environment that supports next-generation
communication models that use the Internet and Internet applications. SIP is
described in IETF RFC 3261 (June 2002), which renders obsolete RFC 2543 (March
1999).
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