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Table Of Contents  CertiGuide to A+ (A+ 4 Real)
 9  Chapter 13: Basic Networking Terminology

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Video and Voice Transmission over IP

After describing of main network technologies, it is impossible to finish this chapter without writing some words about multi service networks.

Video-over-IP and Voice-over-IP implementations enable users to carry video and voice traffic (for example, TV broadcasting and video conference, telephone calls and faxes) over an IP network. There are four main causes for the evolution of the Voice & Video over IP market: low cost phone calls, increasing demand in video conferences for business, add-on services and unified messaging, merging of data/voice/video infrastructures.

A system consists of a number of different components: Gateway, Gatekeeper, Call agent, Call manager, etc.

The Gateway converts media provided in one type of network to the format required for another type of network. For example, a Gateway could terminate bearer channels from a switched circuit network and media streams from a packet network. This gateway may be capable of processing audio, video in any combination, and is capable of full duplex media translations. The Gateway may also play audio/video messages and performs other IVR functions, or may perform media conferencing.

In VoIP, the digital signal processor (DSP) segments the voice signal into frames and stores them in voice packets. These voice packets are transported using IP in compliance with one of the specifications for transmitting multimedia (voice, video, fax and data) across a network: H.323 (ITU), MGCP (level 3, Bellcore, Cisco, and Nortel), MEGACO/H.GCP (IETF), SIP (IETF), T.38 (ITU), SIGTRAN (IETF), Skinny (Cisco) etc.

Coders are used for efficient bandwidth utilization. Different coding techniques for telephony and voice packet are standardized by the ITU-T in its G-series recommendations: G.723.1, G.729, G.729A etc.

The coder-decoder compression schemes (CODECs) enable on both ends of the connection and the conversation proceeds using Real-Time Transport Protocol/User Datagram Protocol/Internet Protocol (RTP/UDP/IP) as the protocol stack.

A number of advanced methods are used to overcome the hostile environment of the IP net and to provide an acceptable Quality of Service. Examples of these methods are delay, jitter, echo, congestion, packet loss, and incorrect order packets arrival. As VoIP is a delay-sensitive application, a well-engineered, end-to-end network is necessary to use VoIP successfully. The Mean Opinion Score is one of the most important parameters that determine the QoS.

Each CODEC provides a certain quality of service. The quality of transmitted speech is a subjective response of the listener (human or artificial means). A common benchmark used to determine the quality of sound produced by specific CODECs is the mean opinion score (MOS). With MOS, a wide range of listeners judge the quality of a voice sample (corresponding to a particular CODEC) on a scale of 1 (bad) to 5 (excellent).

The following are examples of services provided by a Voice over IP network according to market requirements: Phone to phone, PC to phone, phone to PC, fax to e-mail, e-mail to fax, fax to fax, voice to e-mail, IP Phone, toll free number (1-800), class services, call center applications, VPN, Unified

Messaging, Wireless Connectivity, IN Applications using SS7, IP PABX and soft switch implementations.

Thus, constructing such networks can result for the corporation lots of benefits both monetary and productive. In addition, in the near future almost all networks will have in their structure voice and video implementations.

Main directions of development are:

  • new QoS features,

  • IP integrated PBX,

  • Video & Audio CODECs development,

  • Billing and soft switches software.

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