Convergence of voice video and data
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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

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Objectives l.jpg

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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  • 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 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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  • 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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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 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:

    • 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

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

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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.

  • 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.

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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.

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  • 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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  • 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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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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  • 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.