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VoIP - Voice Over IP - Internet Telephony | ||||
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VoIP (Voice over Internet Protocol) is a technology used for sending sound via network based on IP protocol, which is e.g. Internet. It allows for integration of telephony and data transmission into one net thanks to treating sound as an ordinary stream of data. This technology is more and more often perceived as an alternative to casual telephone networks due to its numerous advantages. VoIP enables making phone calls also to subscribers who do not have Internet and use traditional telephone network or a mobile. The basic principle of the technology's operation is relatively simple, namely the speech signal is transferred to digital representation, it undergoes compression with a proper codec and then it is divided into packets and sent via IP protocol. Many new technologies are used here, including logical elements and special protocols. Logical elements are necessary for, among others, managing notifications and storing information on it, routing packets etc. New protocols are mainly signal protocols, which means that they are used for making connections and multimedia sessions, determining user's location, translation of addresses, negotiations of parameters of the notification link, disconnecting and managing notifications, billings and realizations of safety mechanisms. At present two main protocols are used for VoIP, namely H.323 and SIP. H.323 Protocol The first version of the protocol was adopted over 10 tears ago, namely in 1996, the second one in 1998. It belongs to the family of H.32x protocols, which describe multimedia connections inside various networks: H.320 - narrow-band digital networks ISDN H.321 - broad-band digital networks ISDN and ATM H.322 - packet networks with guaranteed band H.323 - packet networks with no guaranteed band H.324 - analogue POTS networks All of these protocols support various sets of audio and video codecs, depending on the band made available in the network. They can operate with transmission control (TCP) or without it (UDP), where in VoIP connections no control protocols are used due to additional delays. H.323 supervises the process of sending multimedia data in packet networks, performing this task in real time. Components of H.323 precisely define how particular components of the system, operating according to the protocol, initiate multimedia sessions and how working posts exchange compressed audio and video data between themselves. H.323 supervises processes of sending multimedia data in packet networks, performing this task in real time. Architecture of telephone IP network, base on H.323 standard, consists of four basic elements: terminals, gatekeepers, gateways and MCU (Multipoint Control Units). Terminals are clients who have the possibility of initiation and receiving notifications. They are also used for sending and receiving two-way stream of data. A terminal can be both, software operating on a PC and a special device designed for this purpose. All terminals should make carrying out a telephone conversation possible, while data or video service is optional. Gatekeepers manage the so-called zone, which is a collection of terminals, gateways and MCU. H.323 standard divides the net into these zones. Notifications inside a net are managed by a gatekeeper. Inter-zone notifications can engaged many gatekeepers. A gatekeeper, when present in the net, supervises the course of all telephone conversations carried out in a zone. Its basic tasks are the following: control of the available band, routing notifications, receiving, refusing notifications in a zone, translation of addresses and user authorization. Gatekeeper is also an interface to other H.323 networks. A "gatekeeper" is an optional element of the network but if it is present in a given subnetwork, then the terminals can use it. A gateway is responsible for connecting the telephone IP network to other types of network. For example, a gateway can connect H.323 network with SIP, PSTN (Public Switched Telephone Network) or ISDN networks. The gateway must provide an interface of real time between various transmission formats and communication procedures. Moreover, it is responsible for establishing and disconnecting connections in both connected networks. The gateway must therefore have mechanisms converting various formats at its disposal and it must operate networks based on various technologies. MCU runs conferences in which at least three end points participate. An MCU unit manages conference resources, runs negotiations between end points (agreeing on, for instance, the method of encoding audio and video data) and it can steer streams of packets containing multimedia data. MCU consists of two basic elements: Multipoint Controller (MC) and optionally a few Multipoint Processors (MP). MC is responsible for exchange of information and for negotiation of communication parameters between the end points, it runs H.255.0 signaling MP is among others responsible for mixing various multimedia data, format translations and eventual redistribution of streams from users. What is very often possible is integration of network elements in one physical device. For example, functionality of a gatekeeper can be combined with functionality of a gateway and an MCU or an MCU can be built into the terminal so as to make conference connections possible without additional devices. See also: Session Initiation Procol in VoIP |
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