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Convergence of Voice,

Video, and Data

Objectives
In this chapter, you will learn to:

Identify terminology used to describe applications and other


aspects of converged networks
Describe several different applications available on converged
networks
Outline possible VoIP implementations and examine the costs and
benefits of VoIP
Explain methods for encoding analog voice or video signals as
digital signals for transmission over a packet-switched network
Identify the key signaling and transport protocols that may be
used with VoIP
Understand Quality of Service (QoS) challenges on converged networks and discuss techniques that can improve QoS

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.

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:
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

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.

VoIP and Traditional Telephones

VoIP and IP Telephones

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 users 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 telephones 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 partys IP
telephone screen.

VoIP and IP Telephones

VoIP and Softphones

VoIP and Softphones

Fax over IP (FoIP)

Fax over IP (FoIP)

Fax over IP (FoIP)

Vidoeconferencing
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.

Call Centers

Call Centers

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


users productivity by minimizing the number of
devices and different methods she needs to
communicate with colleagues and customers.

VoIP Over Private Networks

VoIP Over Private Networks contd

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

VoIP Over Public Networks


To carry packet-based traffic, common carrier networks
incorporate the following:

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.

VoIP Over Public Networks

VoIP Over Public Networks

Softswitch - is a
computer or group of
computers that
manages packetbased traffic routing
and control.

VoIP Over Public Networks

VoIP Over Public Networks

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

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

Waveform Codecs

G.711 - known as a waveform codec because it obtains


information from the analog waveform, and then uses this
information 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). In DPCM, the codec samples the actual
voice signal at regular intervals.

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)


- in 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.

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.

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.

Hybrid Codecs

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.

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.

Media Gateway Control Protocol


(MGCP) and MEGACO (H.248)

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 Controlledload 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.

Differentiated Service (Diffserv)


A technique that addresses QoS issues by
prioritizing traffic.

DiffServ defines two types of forwarding:


Expedited Forwarding (EF)
Assured Forwarding (AF)

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.

Multiprotocol Label Switching

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.