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

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)

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 client’s request.

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 he or 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. 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.

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.

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.

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

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.

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.

Media Gateway Control Protocol (MGCP) and MEGACO (H.248)

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.

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.

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.

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.

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.