Exam tip wan link t carriers t1 t3 and e e1 e3

EXAM TIP
Wan link T carriers (T1, T3) and e (e1, e3) carriers use time-division multiplexing (TDM) to
share the media among multiple systems or customers.
Private Branch exchange (PBx)
Not all information commuting over these various networks is data. With improved perfor -
mance and priority-rating mechanisms such as QoS, digital networks carry time-sensitive
payloads such as voice and video. One such system is the PBX.
The PBX system is a privately (business) owned telephone exchange that connects to a T1
or T3 dedicated line. Each B channel represents another phone extension on the system, or
the unused channels could be aggregated to provide high-speed Internet connectivity.
PBXs have been the target of various attacks and are often overlooked during security
audits and penetration testing. Most PBXs can be remotely configured by way of an ana -
log modem connected to the device’s administrative console. These connections are often
forgotten and left unsecured by strong and changing passwords. Many attacks have occurred
by an attacker dialing in to the administrative console and configuring remote, long-distance
calling accounts. The attacker would sell long-distance service, but the PBX owner would pay
the long-distance charges. The attacker could also interrupt or eavesdrop on phone calls on
the PBX.
Voice over Internet Protocol
VoIP is another WAN technology. By using audio and video codecs (encoder/decoders) to
encode speech into digital data, VoIP places voice (and possibly video) data into the payloads
of IP packets for delivery to the recipient. If the recipient is also on a VoIP system, the packets
will quite likely commute directly to the recipient wholly over the IP network. This can drasti -
cally reduce or eliminate long-distance charges, something that used to take their toll on the
company budget. If the recipient is not on a VoIP phone, the packets will be delivered to a
VoIP media gateway that connects VoIP to many telecommunications systems, such as PSTN,
cellular, and PBX.
Session Initiation Protocol ( SIP ), by the IETF, is commonly used to create, modify, and termi-
nate two-party (unicast) or multiple party (multicast) VoIP sessions. Interestingly, SIP is largely
based on HTTP request and response mechanisms to establish the VoIP calls. A competing
technology to SIP is the H.323 protocol from ITU-T for initiating audio and video sessions over
a network.
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492 CHAPTER 7 Telecommunications and network security
VoIP is not without its share of technological and logistical problems. Audio and video
data is time sensitive and must arrive promptly, or it has lost its usefulness. Because packet-
switched networks, like most IP networks today, can introduce variable delays in the delivery
of packets, VoIP is subject to latency (time lag) and jitter (interruptions in the voice stream). To
overcome this, VoIP relies heavily on QoS features to require priority forwarding of the VoIP
packets through the IP network. The Real-Time Protocol ( RTP ), another IETF protocol, assists
with these time-sensitive communications, which helps improve QoS and session initiation.