Learn 20 Basic Terms of Telecommunication Domain

Learn 20 Basic Terms of Telecommunication Domain

ADSL (Asymmetric Digital Subscriber Line): a DSL option that allows users to transmit data at a speed of up to 24 Mbps, and from the user at speeds up to 3.5 Mbps.

ADSL2 is a modification of the “classical” ADSL technology. They were developed taking into account the increased requirements of providers and end users. In ADSL2 and ADSL 2+, with practically the same transmission range as in ADSL, the speeds are increased to 12 and 25 Mbit / s, respectively. In addition, the adaptive speed change function is implemented. Thanks to these changes, it became possible to support a large number of new applications and additional services (video, multimedia, etc.).

Bitrate (English bitrate) – literally, the speed of passing bits of information. The bitrate is used to measure the effective data rate on the channel, i.e., the rate of transmission of the “useful information” (in addition to that, overhead information, for example, start and stop symbols in asynchronous transmission, or control symbols with redundant coding) can be transmitted over the channel.

APPN (Advanced Peer-to-Peer Networking) is an extension of the SNA architecture that allows the system to automatically adjust to the network topology and send information over the network, bypassing the host computer.

ATM (Asynchronous Transfer Mode) – the technology of high-speed simultaneous transmission of all types of traffic (data, voice, and video) in networks with switched channels; Standard for switched networks, approved by the CCITT in 1985.

DTMF (Dual Tone Multi-Frequency) – two-tone multi-frequency dialing, each digit is transmitted by a combination of two tones.

DWDM (Dense Wave Division Multiplexing, Dense WDM) – a technology of multiplexing in fiber-optic communication lines, is based on the use of light waves of various lengths.

The codec is short for coder/decoder or compressor/decompressor) is a device or program that can perform the conversion of a data stream or signal. Codecs can both encode the stream/signal (often for transmission, storage or encryption) and decode – for viewing or changing in a format more suitable for these operations. Codecs are often used for digital video and audio processing.

G.711 is the ITU-T standard for audio companding. It is mainly used in telephony. First introduced in 1972

G.723 is the voice encoding standard adopted by ITU-T in 1988. The standard describes a broadband codec that converts the audio signal using ADPCM and operates at 24 and 40 kbps.

G.729 is a narrow-band speech codec that is used for the efficient digital representation of narrow-band telephone speech (telephone quality signal). Such speech is characterized by a band between 300 and 3400 Hz and can be digitized with a sampling frequency of 8 kHz. Ideally, a speech codec should represent a speech as wide as possible. In this case, the restored speech will correspond exactly to the original. In practice, it is necessary to choose the bit depth of the codec and to put up with some quantization error.

A router is a network device, based on information about the network topology and certain rules, deciding whether to forward network layer packets of the OSI model between different network segments.

It operates at a higher level than the switch and is more advanced in functionality than the network bridge.

A proxy server is a service in computer networks that allows clients to perform indirect requests to other network services. First, the client connects to the proxy server and requests a resource (for example, a file) located on another server. Then the proxy server either connects to the specified server and receives a resource from it, or returns the resource from its own cache (in cases where the proxy has its own cache). In some cases, the client request or server response can be changed by the proxy server for certain purposes.

IETF (Internet Engineering Task Force) is a public organization that discusses the technical problems of the Internet and creates working groups to solve them.

IP (RFC 791) is used for the unreliable delivery of data (divided into so-called packets) from one network node to another. This means that at the level of this protocol there is no guarantee of reliable delivery of the packet to the addressee. In particular, packets may not arrive in the order in which they were sent, be damaged or not arrive at all. Guarantees of error-free delivery of packets are given by protocols of a higher (transport) level – for example, TCP – which use IP as a transport.

ISDN The primary purpose of ISDN is to transfer data at a speed of up to 64 Kbps over a 4-kilo wired line and provide integrated telecommunications services (telephone, fax, etc.).

Jitter (jitter) – unwanted phase and/or frequency random deviations of the transmitted signal. They arise due to the instability of the master oscillator, changes in the parameters of the transmission line in time, and the different propagation speeds of the frequency components of the same signal.

LAN A local area network (LAN) is a computer network covering a relatively small area, such as a house, office, or a small group of buildings, for example, an institute.

OSI (Open Systems Interconnection) is a seven-level model of data transfer protocols, approved by ISO in 1984 to provide interoperability of open systems.

SIP (Session Initiation Protocol) is an application-level protocol developed by the IETF MMUSIC Working Group and a proposed standard for a method for setting, changing and terminating a user session that includes multimedia elements such as video or voice, instant messages Messaging), on-line games and virtual reality.

SNA (Systems Network Architecture) – the architecture of a computer network for corporate systems, developed by IBM.

TDM (Time-Division Multiplexing) – the technology of combining information coming in several low-speed lines and its further transfer via one high-speed communication channel.

UMTS (Universal Mobile Telecommunications Systems) – the third generation of mobile telephony, will replace the existing GSM systems in the medium term. UMTS provides two main components: a radio network and a carrier network. A radio network consists of mobile equipment and a base station, between which data transmission is switched. The carrier network, in turn, connects the base stations to each other and also creates connections to the ISDN network and the Internet.

VoIP (Voice-over-IP) is a communication system in which an analog audio signal from one subscriber is digitized (encoded) digitally, compressed and sent over digital communication channels to the second subscriber where the reverse operation is performed – Decompression, decoding and playback of an analog signal.

VPN (Virtual Private Network) is a logical network created on top of another network, for example, the Internet. In spite of the fact that communications are carried out through public networks, using unsafe protocols, encryption creates information channels that are closed to outsiders.

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Posted on: June 17, 2017Ana Nichols