Information Technology Reference
In-Depth Information
SIP
SIP is a protocol defined by the IETF and specified in RFC 2543. It is an alternative multi-
media framework to H.323, developed specifically for IP telephony. It is meant to be a
simple lightweight replacement to H.323. Cisco now supports SIP on CUCM, IP phones,
and gateways.
SIP is an application layer control (signaling) protocol for creating, modifying, and termi-
nating IP multimedia conferences, Internet telephone calls, and multimedia distribution.
Communication between members of a session can be via a multicast, a unicast mesh, or a
combination.
SIP is designed as part of the overall IETF multimedia data and control architecture that
incorporates protocols such as the following:
Resource Reservation Protocol (RSVP) (RFC 2205) for reserving network bandwidth
and priority (low-latency) queuing
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RTP and RTCP (RFC 3550) for transporting real-time data and providing QoS feedback
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Real-Time Streaming Protocol (RTSP) (RFC 2326) for controlling delivery of stream-
ing media
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Session Announcement Protocol (SAP) (RFC 2974) for advertising multimedia ses-
sions via multicast
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Session Description Protocol (SDP) (RFC 2327) for describing multimedia sessions
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SIP supports user mobility by using proxy and redirect servers to redirect requests to the
user's current location. Users can register their current locations, and SIP location services
provide the location of user agents.
Figure 14-22 shows SIP components.
CUCM
SIP Proxy
SIP Trunk
IP
SIP
SIP
SIP
SCCP
RTP
SIP Voice Mail
Figure 14-22
SIP Architecture
SIP uses a modular architecture that includes the following components:
SIP user agent (UA):
These endpoints create and terminate sessions, SIP phones,
SIP PC clients, or gateways. A UA client (UAC) initiates a SIP request. A UA server
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CUCM can act as both a server and a client.
