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SWITCHING,ROUTING,ANDSIGNALING 787significance only) as necessary. MPLS represents a sig-
nificant shortcut from the usual IP approach, where every
relay node must look deeply into the packet header, search
a routing table for the best match, and then select the best
next hop toward the packet’s destination. All packets with
the same MPLS label will follow the same route through
the network. In fact, MPLS is designed so that it can ex-
plicitly and flexibly allocate network resources to meet
particular objectives such as assigning the fastest routes
for delay-sensitive packet flows, underutilized routes to
balance traffic better, or multiple routes between the same
end-points for flows with different requirements. This is
calledtraffic engineeringand serves as the foundation for
both optimizing performance and supporting QoS guar-
antees.
Nothing about the MPLS design limits its use to the
IP environment; it can work with suitably equipped ATM
and frame relay routers as well. In fact, it can coexist with
legacy routers not yet updated with MPLS capabilities,
and it can be used in an internetwork that contains a mix
of IP, ATM, and frame relay. Another powerful feature is
the ability to stack labels on a last-in-first-out basis, with
labels added or removed from the stack by each LSR as
appropriate. This allows multiple label-switched paths to
be aggregated into a tunnel over the common portion of
their route for optimal switching and transport. MPLS
is also a convenient mechanism to support virtual pri-
vate networks, especially when multiple Internet service
providers are involved along the path from one end to the
other.Signaling and Interworking
Connection-oriented networks require specific mecha-
nisms for establishing a circuit (physical or virtual) prior
to traffic flow, and for terminating the circuit after-
ward. In the circuit-switched telephony environment, call
setup and termination are part of a well-developed set of
telecommunication system control functions referred to
assignaling. MANs and WANs that were built for voice
included signaling as an integral part of their designs,
because resources were dedicated to each call as it was
established and needed to be released after call comple-
tion.
The ITU-T began developing standards for digital
telecommunication signaling in the mid-1960s; these have
evolved into common channel interoffice signaling system
7 (CCIS7, known in the United States as Signaling System
7, or justSS7for short), currently in use around the world.
SS7 is an out-of-band mechanism, meaning that its mes-
sages do not travel across the same network resources as
the conversations it was designed to establish and control.
In fact, SS7 uses packet switching to deliver control mes-
sages and exchange data, not just for call setup, but also
for special features such as looking up a toll-free number
in a database to find out its real destination address, call
tracing, and credit card approvals. Out-of-band delivery
of the messages allows SS7 to be very fast in setting up
calls, to avoid any congestion in the transport network,
and also to provide signaling any time during a call.
The SS7 network has a number of elements that work
together to accomplish its functions (Figure 13):STPSTPSCPSSPSSPdata
transport
networkSS7
networkDB(^123456789)
8#
(^123456789)
8#
Figure 13: SS7 network elements.
Signal switching points (SSPs) are the network edge de-
vices responsible for setting up, switching, and termi-
nating calls on behalf of connected subscriber devices,
and thus insert user traffic into, and remove it from,
the service provider’s backbone network.
Signal transfer points (STPs) are packet switches respon-
sible for getting SS7 messages routed through the
control network.
Signal control points (SCPs) house the databases that
support advanced call processing.
In packet-switched MANs and WANs, signaling had
been associated primarily with establishing and tearing
down SVCs that required no further control during the
data transfer phase. With a rising interest in multime-
dia communications (e.g., video, and especially voice over
IP) however, the ITU-T quickly recognized a need for ad-
ditional capabilities. TheirH.323 recommendations en-
compass an entire suite of protocols that cover all aspects
of getting real-time audio and video signals into packet
form, signaling for call control, and negotiation to ensure
compatibility among sources, destinations, and the net-
work. H.323 takes advantage of prior ITU work (such as
ISDN’s Q.931 signaling protocol) and defines four major
elements (Figure 14):
Terminalsare the end-user devices that originate and re-
ceive multimedia traffic.
Gatewaysprimarily handle protocol conversions for par-
ticipating non-H.323 terminals, as would be found in
the public switched telephone network (PSTN).
Gatekeepersare responsible for address translation, call
control services, and bandwidth management.
Multipoint Control Units(MCUs) provide multiconferenc-
ing among three or more terminals and gateways.
The IETF took a simpler approach to signaling with the
session initiation protocol(SIP), which was designed as a
lightweight protocol simply to initiate sessions between
users. SIP borrows a great deal from the hypertext trans-
fer protocol (HTTP), using many of the same header fields,