Convergence of Voice, Video, and Data Chapter 14

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

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

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

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.

IQ TEST 1 1. _____ uses packet-switched networks to transmit faxes from one node on the network to another. Answer: Fax over IP (FoIP) 2. True or False: Completing calls over a packet-switched network is known as toll bypass. Answer: True 3. Gateways used on converged networks may also be called _____. Answer: media gateways 4. A computer programmed to act like an IP telephone is known as a(n) _____. Answer: softphone 5. What term is used to describe the process of issuing real-time video signals from a server to a client? Answer: Video streaming

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

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

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 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 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 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 the H.323 protocol that handles call signaling. H 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

IQ TEST 2 1. _____ means that the network attempts to deliver packets in a timely fashion and in the proper sequence, but does not guarantee either. Answer: Best effort 2. A(n) _____ controller is a computer that manages multiple media gateways. Answer: media gateway 3. A(n) _____ is a computer or group of computers that manages packet- based traffic routing and control. Answer: softswitch 4. On the PSTN, a telephone switch accomplishes the coding and decoding, or codec, by using a computer chip known as a(n) _____. Answer: digital signal processor (DSP) 5. _____ do not attempt to replicate a voice signal’s original waveform Answer: Vocoders

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.