Convergence of Voice, Video, and Data

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Convergence of Voice, Video, and Data

• Chapter 14

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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 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
• 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 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.

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VoIP and IP Telephones
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VoIP and Softphones
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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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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. Used with point-to-multipoint
  video.
• Broadcast video server issues separate copies
  of the video signal to every client, upon the
  clients request.

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

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VoIP Over Private Networks
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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

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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 end-users.

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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 packet-based 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 information from the analog waveform, and
  then uses this information to reassemble the
  waveform as accurately as possible at the
  receiving end. This does not manipulate the
  signal in any way. It simply tries to
  reconstruct it. This is a concern in
  packet-based networks. Requires 64 Kpbs.
  Requires a significant amount of throughput.

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Waveform Codecs

• 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. Is able to predict voice samples, so
  required 6.4 Kbps, only one-tenth of G.711. Not
  as good voice quality but adequate for
  packet-based networks using VoIP and
  videoconferencing.

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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) - 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. The
  result is more accurate predictions.
• G.726 uses ADPCM and can operate over a 16-, 24-,
  32-, or 40 Kbps channel.

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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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VoIP Signaling Protocols

• 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. Released after H.323 and
  has never received as much usage.
• 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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VoIP Transfer Protocols

• Real-Time Transport Protocol (RTP). Operates on
  top of UDP at the Transport Layer of OSI model.
  Indicates what order packets should be assembled
  by assigning each packet a time stamp. Cannot do
  anything to correct transmission flaws.

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Quality of Service (QoS)

• Resource Reservation Protocol (RSVP) A QoS
  technique that attempts to reserve a specific
  amount of network resources for a transmission
  before the transmission occurs. Emulates a
  circuit-switched connection.
• Allows for two service types Guaranteed service
  (will not suffer packet loss and minimal delay)
  and Controlled-load service expected transmission
  if network carried little traffic).
• 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. Adds information in Type
  of Service field in an IP version 4 datagram.
  (See Chapter 7).
• DiffServ defines two types of forwarding
• Expedited Forwarding (EF) minimum departure rate
• Assured Forwarding (AF) different levels of
  router resources assigned to data streams.

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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.
  No strictly a QoS technique but rather a way of
  forwarding packets.
• To indicate where data should be forwarded,
  Multi-protocol Label Switching (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. With
  MPLS, data streams are more likely to arrive
  without delay.

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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.