VoIP Technology

VoIP protocols and technologies are divided into two major categories: Distributed and Centralized. The MGCP protocol is an example of Centralized VoIP protocol since it supports a centralized client / server architecture. On the other hand, H.323 and SIP protocols are categorized as Distributed since the voice distribution is based on inter-node ad hoc network. All VoIP technologies and protocols use a common protocol for speech distribution which is RTP (Real Time Protocol) which packetizes voice traffic over IP, as well as supports multiple codecs to compress the data.The differences lies in the way of signal transmission and in the area of service logic and mode of the call management (at the end points or on a central server). Both architectures (Distributed and Centralized) have their advantages and disadvantages. Distributed models scale well and are more flexible (robust) because they have no central node, which can lead to failure. Centralized model are more easily managed and support the traditional supplementary services (such as conferences), but may have limits on scalability which determines the capacity of the central telephony server.

IP telephony has to do mainly with digital telephony systems (LAN based IP PBX systems) which use the IP protocol entirely for voice communication. All components of the IP telephony system use digitized voice which is transferred as IP packets through an IP network (usually the LAN network). The telephone handsets (VoIP phones) translate the analogue voice signal into digital voice (binary voice) which is transferred as IP packets from one phone to another. The call control system is usually a software based (softswitch) server which handles all call signaling, call routing, IP phone management etc, again using IP protocol for transport. So think about IP telephony as a bigger concept. VoIP on the other hand is a subset of IP Telephony. Basically, VoIP is the technology which is used as the vehicle to transport phone calls. VoIP is the technology in which the analogue voice signal is digitized (analog to digital conversion) and becomes binary numbers in order to be transferred by the IP protocol.

A typical case is the use of ADPCM coding ITU-T G.726 with a 4-bit quantization, providing a transfer speech rate of 32 kbit / sec. Unlike PCM, the 4-bit code is not the actual amplitude of the speech, rather it is the difference in the amplitude and the rate of change of amplitude, using some rather primitive linear prediction. The new methods of compression, such as LPC, CELP, and MP-MLQ, use additional features in the waveform in both PCM and ADPCM, using knowledge of the original features of the formation of speech. Such methods are applied methods of signal processing, which compress speech by sending only simplified parametric information about the original form of sound and the vocal tract. To send this information requires less bandwidth. These methods can be combined into a common set of codecs in the source.

Bandwidth requirements are critical and are determined not only by the transmission rate of the codec used (from 3-4 to 64 Kbit / sec), but the extra load on the network, called IP headers, and other factors. Due to the presence of pauses during conversations, a technology was developed for detecting voice activity (Voice Activity Detection, VAD). With VAD, bandwidth requirements are reduced roughly in half. Thus, for example, for the G.711 codec with bandwidth of 64 kbit / sec, with the use of VAD technology the total bandwidth for a voice channel will be about 40 kbit / sec.

IP Telephony systems are those using entirely IP packets for voice communication, as explained before. In contrast to packet switched telephone systems (those based on IP protocol), conventional telephone systems apply the logic of direct connection between the two communicating voice parties through a dedicated circuit reserved exclusively for each contact. Thus the term Circuit switched telephone systems. In packet switched systems, however, the same communication line can be used to simultaneously pass different kinds of packets. Thus, the voice packets of one or more conversations may travel through the same route as other packets transferring data, video etc. This is the main difference between traditional telephony which is implemented to the public switched telephone network (PSTN) and telephony implementation on IP networks (or more generally to packet switched networks).

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