Information Technology Reference
In-Depth Information
Q.931 is a standard for call signaling and setup.
H.225 specifies messages for call control that includes signaling between end point,
registration, admissions, and packetization/synchronization of media streams.
RTP is the transport layer protocol that transports VoIP packets. RCTP is a session layer
protocol.
H.323 includes a series of protocols for multimedia that are listed in Table 12-3.
H.323 Protocols
Table 12-3
Video
Audio
Data
Transport
H.323 Protocol
H.261
H.263
G.711
G.722
G.723.1
G.728
G.729
T.122
T.124
T.125
T.126
T.127
RTP
H.225
H.235
H.245
H.450.1
H.450.2
H.450.3
X.224.0
SIP
SIP is defined by the Internet Engineering Task Force (IETF) and is specified in RFC 2543. It
is an alternative multimedia framework to H.323, developed specifically for IP telephony.
SIP is an application layer control (signaling) protocol for creating, modifying, and terminating
Internet multimedia conferences, Internet telephone calls, and multimedia distribution. Com-
munication between members in a session can be through a multicast, unicast mesh, or a
combination of these.
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 resources
•
RTP (RFC 1889) for transporting real-time data and providing QoS feedback
•
RTSP (RFC 2326) for controlling delivery of streaming media
•
Session Announcement Protocol (SAP) for advertising multimedia sessions through
multicast
•
Session Description Protocol (SDP) (RFC 2327) for describing multimedia sessions.
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