The article contains technical information about the General principles of voice on the Internet and ways of providing telephone service .
The technology of IP telephony (Voice over Internet Protocol - VoIP) provides the possibility of transmission of telephone calls over the Internet, i.e. it is technology for delivery of voice traffic in real-time. Is this data networks using transport mechanisms that operate the packages. The main advantage of the technology is. that its application allows not only to reduce costs but also to combine in a single network transfer voice and data. Used compression algorithms are considerably reduce its volume, and thereby more efficiently use the bandwidth transmittance.
In IP telephony apply two basic circuit connection. The first one provides that all necessary software and hardware, manufacturing digitization, compression, packetization and playback of the audio signal, must be installed on personal computers. In this case, for communication subscribers need not tepatnya devices, and headphones with microphone. For communication only needs the IP addresses of the computers of subscribers.
When the second communication scheme, the user places a call through conventional telephone apparatus, and for the PSTN connection to the Internet using special equipment - telephone the gateway (Internet Telephony Gateway). This equipment converts the analog form digital data, organizes it into a sequence of packets in the format The Internet transmits them to the network, and receives the return packet, converting them in analog form. In addition, the telephony gateway provides the implementation interface with the PSTN, the generation of signals subscriber signalling, coupling and the separation of subscribers.
In IP telephony is paramount to a highly reliable connection, able to support voice communications in real time. This means that transmission from the subscriber to the subscriber should occur with minimal delay.
Modern telephone systems use digital transmission and switching. Between aware, however, that the subscriber line, usually analog. Therefore, here you need the appropriate transformation of signals. Usually it is produced by a local telephone switching station, but this function may to perform a service that is closer to the user, for example, if the digital transmission is used in the local access network.
Transmission over a telephone line is based on the simple digitization of speech with the use of standards developed in the 60-ies. For the speech signal in the band from 300 Hz to 3.1 kHz discretization is applied at a rate of 8,000 samples per second and 8-bit encoding of each frame. Thus, each speech the channel requires 64 kbit/s. Thirty of these voice channels are combined with channels framing and signaling, creating standard digital stream 2048 kbps, which is generally called a 2 Mbps stream, or E1.
Transmission delay in an existing PSTN are very small, they arise in mainly in compression of speech and the use of satellite lines. If the delay exceed 400 MS, this significantly degrades the sound quality.
The development of powerful methods for speech compression reduces the required frequency band. It important for systems where the cost of bandwidth is very high, for example, in a cellular radio. In addition, there is no need to transmit a signal during speech pauses when one person listens to another. The transmission speed of 5 kbps provides the same quality of speech, and standard digital lines with a total speed about 2.5 kbps subject to the intervals of silence. The algorithms employed compression-based discretization and comparing neighboring samples of speech, so introduces some latency. For best speech recognition is also required effective echo cancellation. Use a packet, such as IP transfer, ideal for transfer of speech in this form, provided that in processing included a number of features.
The total transmission delay between subscribers should be kept within certain limits. It is also necessary to minimize the fluctuation of the delay, otherwise the quality reconstructing speech will deteriorate. Latency compression of speech need be taken into account in the overall transmission delay. You need to take special measures to data services in the band tonal frequencies, for example, in relation to modem or Fax traffic, since the compression algorithms of speech are not well suited for the processing of these signals. Any, even minor irregularities in the transmission inevitably felt on the receiving side, leads to distortion of the transmitted information.
However, since we are talking about data transmission system, the signals mentioned above services can be passed on to a more suitable way. There is also need special measures in order to ensure high-quality transmission of any the necessary signals with fixed frequencies. It is mainly related signals tone dialling DTMF phone subscription. Even if these signals are not used when dialing, over time they will often be able to apply, will be involved when the automatic processing system of the call.
The Internet was created in the calculation for packet data transmission between computers where not so important delays in the transmission and the relative delay between packages. Because computers are based on internal processing in under the arrangements of the queue, they can process packets that are the receipt. Differ and mechanisms for routing in the PSTN and the Internet. So by the way, there are some differences in how packets are transmitted and how they routed. These circumstances should be taken into account when considering the the question of how to use the Internet for voice services. It should be remembered that the network designed for data systems. For example, if your e-mail takes a few hours for delivery of information, it is quite acceptable for the recipient.
When solving problems associated with the transmission of speech, in Internee adopted a strategy the provision of services with different quality Grades of Service), in accordance with which certain packets (in this case those associated with the transmission speech) are processed in priority order. For this, of course, users The Internet will have to pay an additional fee.
The Internet is an undefined mass of equipment used, increase which will continue. For distribution of new services on currently standards are already being developed, but progress will be associated with the modernization of equipment suppliers. For example, the specifications for cable modems include detailed requirements on the levels of service quality (Quality of Service-QoS). Funds are also being developed that will allow operators to measure actual levels of service. It should be noted here, what are the requirements to improve its quality in terms of developing The Internet. Tougher they are, and with the advent of special equipment for quality control connection. Already designed monitor to monitor the use of network resources (if doing multiple providers). On the market of equipment appeared and network management tools for alignment traffic and its redistribution between the fiber optic paths.
Most users IP telephony is known, for example, such application, as the provision of intercity and international telephone communication. It is expected that in the future will increase popularity and other applications, in particular one that will allow subscribers to receive phone calls while browsing the Internet. When a call arrives at a busy room PBX forwards it to the gateway IP telephony, which will bring the call to the subscriber at IP network, and special software will be reported. The subscriber in this case resolves itself, to answer the call or not. For widespread deployment of such services in Russia needs to solve one problem, but it's pretty serious: all the terminal the station must support the forwarding of the call.
Author: K. Sanvik, London, UK