IP telephony. Encyclopedia of radio electronics and electrical engineering

Encyclopedia of radio electronics and electrical engineering / Telephony

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The article provides technical information about the general principles of voice communication over the Internet and methods of providing telephone services.

IP telephony technology (Voice over Internet Protocol - VoIP) provides the ability to transmit telephone calls over the Internet, i.e. it is a technology for delivering voice traffic in real time. This is done over data networks using transport mechanisms that operate on packets. The main advantage of the technology is that... that its use allows not only to reduce costs, but also to combine voice and data transmission in one network. The information compression algorithms used in this case significantly reduce its volume, and thus the bandwidth is used more efficiently.

IP telephony uses two basic communication schemes. The first of them stipulates that all the necessary software and hardware that perform digitization, compression, packetization and playback of an audio signal must be installed on personal computers. In this case, for communication between subscribers, not telephones are required, but headphones with a microphone. To establish a connection, only the IP addresses of subscriber computers are needed.

In the second communication scheme, the user calls using a regular telephone, and special equipment is used to connect the PSTN to the Internet - an Internet Telephony Gateway. This equipment converts the analog form of information into digital, organizes a sequence of packets from it in Internet format, transmits them to the network, and also receives return packets, converting them into analog form. In addition, the telephone gateway provides the implementation of an interface with the PSTN, generation of subscriber signaling signals, connection and disconnection of subscribers.

In IP telephony, a highly reliable connection that can support real-time voice transmission is critical. This means that transmission from subscriber to subscriber should occur with minimal delays.

Modern telephone systems use digital transmission and switching. Meanwhile, it is known that subscriber lines are usually analog. Therefore, appropriate signal conversion is necessary here. This is usually carried out at the local telephone switching office, but this function may also be performed by a service located closer to the user, for example if digital transmission is used in a local access network.

Telephone transmission is based on simple speech digitization using standards developed in the 60s. For speech signal in the band from 300 Hz to 3,1 kHz, sampling is used at a rate of 8000 samples per second and 8-bit encoding of each sample. Thus, each voice channel requires 64 kbit/s. Thirty of these voice channels are combined with frame synchronization and signaling channels to form a standard 2048 kbit/s digital stream, which is usually called a 2 Mbit stream, or E1.

Transmission delays in the existing PSTN are very small, and they arise mainly during speech compression and the use of satellite lines. If delays exceed 400 ms, the sound quality will noticeably deteriorate.

The development of powerful speech compression techniques reduces the required frequency bandwidth. This is essential for systems where the cost of frequency bandwidth is very high, for example, in cellular radio communications. In addition, there is no need to transmit a signal during speech pauses when one subscriber is listening to another. A transmission rate of 5 kbit/s provides the same speech quality as standard digital lines, with a total speed of about 2,5 kbit/s including silent intervals. The compression algorithms used are based on sampling and comparison of adjacent speech samples, so some delay is introduced. Better speech recognition also requires effective echo cancellation. The use of packet transmission, such as IP transmission, is ideal for transporting speech in this form, provided that a number of features are included in the processing.

The overall transmission delay between subscribers should be kept within certain limits. It is also necessary to minimize delay variations, otherwise the quality of the reconstructed speech will deteriorate. The processing delay of speech compression must be factored into the overall transmission delay. Special considerations are required for voiceband data services, such as modem or fax traffic, as speech compression algorithms are not well suited to processing these signals. Any, even minor, disturbance in transmission is inevitably felt on the receiving side and leads to distortion of the transmitted information.

However, since we are talking about a data transmission system, the signals of the above-mentioned services can be transmitted in a more suitable way. Here, too, special measures are required to ensure high quality transmission of any required fixed frequency signals. This mainly concerns DTMF tone dialing signals from the subscriber telephone. Even if these tones are not currently used in dialing, over time they may be used frequently when automated call handling systems are deployed.

The Internet was created with the expectation of packet data transfer between computers, where transmission delays and relative delays between packets are not so important. Because computers rely on internal processing based on queuing mechanisms, they can process packets as they arrive. Routing mechanisms in the PSTN and on the Internet also differ. Thus, there are differences in how packets are transmitted and how they are routed. These circumstances must be taken into account when considering how to use the Internet for voice services. Remember that the network is designed for data systems. For example, if an email takes several hours to deliver information, then this is quite acceptable to the recipient.

When solving problems associated with voice transmission, the Internet has adopted a strategy for providing services with different quality (Grades of Service), according to which certain packets (in this case, those associated with voice transmission) are processed in priority order. For this, of course, Internet users will have to pay an additional fee.

The Internet represents an indefinite mass of usable equipment, which will continue to increase. Standards are already being developed to distribute new services, but progress will still depend on the modernization of equipment by its suppliers. For example, specifications for cable modems include detailed requirements for Quality of Service (QoS) levels. Tools are also being developed to allow operators to measure actual service levels. It should be noted here that the requirements for improving its quality are related to the further development of the Internet. They are also becoming stricter with the advent of special equipment for monitoring the quality of communications. A monitor has already been developed that allows you to monitor the use of network resources (if several providers do this). Network management tools for traffic equalization and redistribution between fiber-optic paths have also appeared on the equipment market.

Most users of IP telephony know, for example, such an application as providing long-distance and international telephone communications. It is expected that other applications will also increase in popularity in the future, in particular one that will allow subscribers to receive phone calls while surfing the Internet. When a call arrives at a busy number, the PBX forwards it to the IP telephony gateway, which will transfer the call to the subscriber via the IP network, and special software will notify him about this. In this case, the subscriber decides for himself whether to answer the call or not. For widespread use of such a service in Russia, one problem must be solved, but it is quite serious: all end stations must support the call forwarding function.

Author: K. Senwick, London, UK

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