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    Voip Voip Presentation Transcript

    • 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 net-works 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 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.
    • 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 user’s 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 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
    • 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.
    • VoIP Over Public Networks
    • VoIP Over Public Networks
      • Softswitch - is a computer or group of computers that manages packet-based 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 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.
    • 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.