ch 14 Convergence of Voice, Video, and Data

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Transcript ch 14 Convergence of Voice, Video, and Data

The Convergence of:
Voice, Video, and Data
Objectives
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Identify terminology used to describe applications and other aspects of
converged networks
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Describe several different applications available on converged networks
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Outline possible VoIP implementations and examine the costs and benefits
of VoIP
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Explain methods for encoding analog voice or video signals as digital
signals for transmission over a packet-switched network
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Identify the key signaling and transport protocols that may be used with
VoIP
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Understand Quality of Service (QoS) challenges on converged net-works
and discuss techniques that can improve QoS
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Terminology
Voice over IP (VoIP) - the use of any network (either public or
private) to carry voice signals using TCP/IP.
Voice over frame relay (VoFR) - the use of a frame-relay network
to transport packetized voice signals
Voice over DSL (VoDSL) - the use of a DSL connection to carry
packetized voice signals
Fax over IP (FoIP) - uses packet-switched networks to transmit
faxes from one node on the network to another.
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Voice Over IP (VoIP)
The use of packet-switched networks and the TCP/IP protocol
suite to transmit voice conversations.
Reasons for implementing VoIP may include:
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To improve business efficiency and competitiveness
To supply new or enhanced features and applications
To centralize voice and data network management
To improve employee productivity
To save money
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VoIP and Traditional Telephones
Techniques for converting a telephone signal from digital
form include:
Using an adapter card within a computer workstation.
Connecting the traditional telephone to a switch capable of accepting
traditional voice signals, converting them into packets, then issuing
the packets to a data network.
Connecting the traditional telephone to an analog PBX, which then
connects to a voice-data gateway to convert the signals.
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VoIP and Traditional Telephones
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VoIP and IP Telephones
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VoIP and IP Telephones
Popular features unique to IP telephones include:
Screens on IP telephones can act as Web browsers, allowing a user to
open HTTP-encoded pages and, for example, click a telephone
number link to complete a call to that number.
IP telephones may connect to a user’s personal digital assistant (PDA)
through an infrared port, enabling the user to, for example, view his
phone directory and touch a number on the IP telephone’s LCD
screen to call that number.
If a line is busy, an IP telephone can offer the caller the option to leave
an instant message on the called party’s IP telephone screen.
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VoIP and IP Telephones
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VoIP and Soft phones
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Soft phone
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VoIP and Softphones
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Fax over IP (FoIP)
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Fax over IP (FoIP)
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Fax over IP (FoIP)
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Video Conferencing
The real-time transmission of images and audio
between two locations.
Video streaming - the process of issuing real-time video
signals from a server to a client.
Video terminals - devices that enable users to watch, listen,
speak, and capture their image.
Multipoint control unit (MCU) - also known as a video
bridge, provides a common connection to several clients.
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Call Centers
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Call Centers
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Unified Messaging
A service that makes several forms of
communication available from a single user
interface.
The goal of unified messaging is to improve a
user’s productivity by minimizing the number of
devices and different methods she needs to
communicate with colleagues and customers.
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VoIP Over Private Networks
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VoIP Over Private Networks cont’d
Characteristics that make a business particularly well-suited to
running VoIP over a private network include:
A high number of telephone lines (for example, more than 100)
Several locations that are geographically dispersed across long distances
(for example, over a continent or across the globe)
A high volume of long-distance call traffic between locations within the
organization
Sufficient capital for upgrading or purchasing new CPE, connectivity
equipment, LAN transmission media, and WAN links
Goals for continued network and business expansion
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VoIP Over Public Networks
To carry packet-based traffic, common carrier networks
incorporate the following:
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Access service - provides endpoints for multiple types of incoming
connections.
Media gateway service - Translates between different Layer 2 protocols
and interfaces.
Packet-based signaling - Provides control and call routing.
Signaling gateway service - Translates packet-based signaling protocols
into SS7 signaling protocol and vice versa.
Accounting service - Collects connection information, such as time and
duration of calls, for billing purposes.
Application service - Provides traditional telephony features to endusers.
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VoIP Over Public Networks
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VoIP Over Public Networks
Softswitch - is a computer
or group of computers
that manages packetbased traffic routing and
control.
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VoIP Over Public Networks
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VoIP Over Public Networks
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Cost-Benefit Analysis
The major costs involved in migrating to and supporting a
converged network include:
Cost of purchasing or upgrading CPE, connectivity devices and
transmission media for each location
Cost of installation services and vendor maintenance
Cost of training technical employees and other staff
Recurring cost of new or expanded connections
Cost of transmitting voice and data, if part of the connection fees are
usage-based
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Cost-Benefit Analysis
Potential economic gains of converged network can be
estimated by taking into account the following:
Bypassing common carriers to make long-distance calls, thus avoiding
tolls
Consolidating traffic over the same connections, which leads to
reducing or canceling PSTN or leased-line connections
Providing employees with more efficient tools and means of
communication
Increased productivity for mobile employees
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Waveform Codecs
G.711 - known as a waveform codec because it obtains signals
from the source, an analog waveform, and then uses the signals
to reassemble the waveform as accurately as possible at the
receiving end.
G.723 - uses a form of PCM known as differential pulse code
modulation (DPCM). Using DPCM, the codec samples the
actual voice signal at regular intervals.
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Waveform Codecs
DPCM codecs - work well with human speech because, within
very short time spans, our speech patterns are predictable.
Adaptive differential pulse code modulation (ADPCM) – with
this codec, not only do the nodes base predictions on
previously-transmitted bits, but they also factor in human
speech characteristics to recreate wave-forms.
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Vocoders
Apply sophisticated mathematical models to voice
samples, which take into account the ways in which
humans generate speech.
G.729 - reduces its throughput requirements by suppressing the
transmission of signals during silences.
 Can operate over an 8-Kbps channel.
 Requires only moderate DSP resources and results in only
moderate delays.
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Hybrid Codecs
Incorporate intelligence about the physics of human
speech to regenerate a signal.
Hybrid codecs use lower bandwidth than waveform
codecs, but provide better sound quality than
vocoders.
One example of a hybrid codec is specified in the ITU
standard G.728.
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Hybrid Codecs
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H.323
An ITU standard that describes not one protocol, but
an entire architecture for implementing multiservice
packet-based networks.
H.225 - the H.323 protocol that handles call signaling.
H.245 - ensures that the type of information, whether
voice or video, issued to an H.323 terminal is formatted
in a way that the H.323 terminal can interpret.
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Session Initiation Protocol (SIP)
SIP was codified by the IETF (in RFC 2543) as a set of
Session-layer signaling and control protocols for
multiservice, packet-based networks.
Because it requires fewer instructions to control a call,
SIP consumes fewer processing and port resources
than H.323.
SIP and H.323 regulate call signaling and control on a
VoIP network. However, they do not account for
communication between media gateways.
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Media Gateway Control Protocol
(MGCP) and MEGACO (H.248)
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Resource Reservation Protocol
(RSVP)
A QoS technique that attempts to reserve a specific
amount of network resources for a transmission
before the transmission occurs.
Allows for two service types: Guaranteed service and Controlled-load
service.
As a result of emulating a circuit-switched path, RSVP provides excellent
QoS.
Because it requires a series of message exchanges before data
transmission can occur, RSVP consumes more network resources
than some other QoS techniques.
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Differentiated Service (Diffserv)
A technique that addresses QoS issues by
prioritizing traffic.
DiffServ defines two types of forwarding:
 Expedited Forwarding (EF)
 Assured Forwarding (AF)
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Multiprotocol Label Switching
Offers a different way for routers to determine the
next hop a packet should take in its route.
To indicate where data should be forwarded, MPLS replaces the
IP datagram header with a label at the first router a data stream
encounters.
The MPLS label contains information about where the router
should forward the packet next.
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Multiprotocol Label Switching
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Summary
VoIP can improve efficiency and competitiveness,
supply new or enhanced features and applications, and
centralize voice and data network management.
Fax over IP (FoIP) is commonly implemented according
to either the ITU T.37 or T.38 standard.
Call centers are good candidates for converged
networks.
Codecs convert analog voice signals into digital form.
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Convergence of Voice,
Video, and Data
End