IP telephony: modern technology of free calls
Introduction
Since the invention of the telegraph, the history of telecommunications goes back more than 200 years. The rapid development of telephony has led to the fact that communication capabilities, which seemed quite fantastic recently, are already being used everywhere today. One of these innovations is IP telephony. Not so long ago, it would be extremely difficult to imagine voice transmission not through the usual telephone wires, but through data transmission channels. Today, telephone networks and the Internet are combined into a single infrastructure. Voice IP traffic has already pressed the usual telephony and is competing with cellular communication. With the development of IP telephony, its functionality is increasing, and to the absence of roaming on the Internet, free audio-video communication, the possibility of holding conferences, cloud services, and so on are added. Very soon, the full potential of IP-telephony will become an integral part of our life, like a TV remote control or an alarm system on a car, but until that happens, I would like to tell you about how IP telephony is implemented and what technologies are used for this in MTT.
Protocols
IP telephony is based on several standardized communication protocols. The underlying protocol is the H.323 standard from the International Telecommunication Union (ITU), according to which transnational Internet telephony operators operate. All calls in this protocol are controlled by the Gatekeeper (gatekeeper, zone controller) - a program or device that is located at the IP-telephony provider that coordinates all clients that are united in a common zone. It manages terminals, gateways, conference management devices (MCUs), etc. Thanks to the gateways, the H.323 protocol can integrate telephone and IP networks. So, the gateway can be an adapter for analog phones (eg HandyTone, GXW), which connects to the Internet and an analog phone. When dialing a number on a traditional telephone, the gateway transfers the call to the Internet, and when a call comes in via IP telephony, it allows the owner of an analog device to receive a call without using the public switched telephone network (PSTN). The gateway can be either special IP phones (manufacturers: Gigaset, Panasonic, Cisco, or Yealink), or programs on a computer to which a headset with headphones and a microphone is connected. Terminals are the end devices from which a call is made: smart phones and PCs with special software, IP phones. For a conference in which more than 3 people can participate, the MCU server is used.
The SIP protocol is considered to be more flexible and scalable, which is used by almost all IP-telephony providers, allowing not only to talk over the Internet, but also to exchange video, instant messages, play online games, etc. Incidentally, the YouMagic IP-telephony service from MTT is built on it, by the way. This protocol regulates only the procedure for establishing a connection between devices, and a network streaming data transfer protocol (SDP) is connected for data transfer. SIP has more functions than H.323, and this difference is due to the fact that this protocol was not developed by ITU telephonists, but by an open group of the international community of designers, scientists, network operators and providers (IETF). SIP protocol requires proxy servers that are installed with IP telephony providers and user terminals, that is, devices through which communication takes place. They can be the same IP phones, gateways, as well as computers or smartphones with installed software (softphone) and connected to the Internet.
There is also a corporate SCCP protocol that works according to the H.323 logic. SCCP was developed by Cisco to reduce communications costs in companies by saving on hardware.
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Codecs
For the quality of voice transmission in all protocols, codecs are responsible - algorithms for converting voice into the required digital format. In IP-telephony, voice is converted into so-called packets, which are transmitted over Internet networks. For voice transmission in good quality, without delay, echo and distortion, turning the voice of a person into the voice of a robot, one or another codec is selected, depending on the Internet connection, program settings and other factors. For example, a single codec (for example, g722) with a fast and confident Internet connection can transmit voice in high quality, but the low bandwidth of the channel will make the conversation impossible due to the loss of phrases or sounds. In MTM's YouMagic, codecs such as iLBC or Silk can transmit voice without noticeable delays when the quality of the Internet channel is low, but they compress the data so much that some voice features are simply lost when encoding-decoding, and even at high speed they are still will not be able to provide high quality dialogue.
SIP Technology Service - YouMagic
Consumers have long been accustomed to comfortable hearing when communicating through fixed networks and even through networks of mobile operators, so any interference and failures in IP-telephony can discourage the desire to use it. Being well aware of this, each Internet telephony operator, when developing its products, has to closely monitor the status of its proxy servers and other equipment, data passing through the networks of other operators and other costly issues, while striving to keep low rates for communication with the whole world.
One of the clearest examples of maximizing the benefits of SIP is YouMagic, launched in 2011. In this purely Russian IP telephony project, agents can be smartphones, personal computers, IP / DECT phones, or regular phones, complete with a special gateway, as discussed above. For common operating systems, their own softphones have been released, allowing you to talk through YouMagic on an iPhone or Android, on a Windows or Mac OS computer.
To ensure good quality of conversation, YouMagic uses different codecs in different situations. The low quality of the Internet channel is typical for the domestic mobile Internet, therefore, when developing an application for smartphones, all the features of packet transmission in 3G networks were taken into account. So, in the YouMagic application for Android OS, automatic selection of the desired codec is provided depending on the Internet connection. And experienced users from the list of presented codecs can independently select those that will be used in certain Internet networks. This feature allows the application to automatically adjust to the properties of the Internet connection, improving the quality of voice transmission in the dialogue.
One of the most convenient types of Internet access, using IP-telephony, is Wi-Fi. Today, Wi-Fi coverage is becoming increasingly widespread, and quality is more stable. Thus, it was recently announced that in 2013 work will begin on creating a Wi-Fi network on all branches of the metro in Moscow, and ground transportation will be connected. Wi-Fi is available in cafes and restaurants in many cities of Russia. It seems that after a couple of years we will be able to stay in the “Wi-Fi cloud”, communicating with the whole world in any way convenient for us, and IP-telephony will become one of the most comfortable ways of communication.
Traditionally, communication between users of one IP-telephony operator is free of charge, and a call to a mobile or landline number in any country or city in the world is much cheaper than using a mobile or landline phone. YouMagic provides its subscribers with extremely affordable rates: Russia will have to spend less than 1 ruble per minute to communicate with any city; calls to such popular countries of the world as the USA, Germany or Israel are also profitable. And communication of all YouMagic owners among themselves, of course, remains free.
Since the invention of the telegraph, the history of telecommunications goes back more than 200 years. The rapid development of telephony has led to the fact that communication capabilities, which seemed quite fantastic recently, are already being used everywhere today. One of these innovations is IP telephony. Not so long ago, it would be extremely difficult to imagine voice transmission not through the usual telephone wires, but through data transmission channels. Today, telephone networks and the Internet are combined into a single infrastructure. Voice IP traffic has already pressed the usual telephony and is competing with cellular communication. With the development of IP telephony, its functionality is increasing, and to the absence of roaming on the Internet, free audio-video communication, the possibility of holding conferences, cloud services, and so on are added. Very soon, the full potential of IP-telephony will become an integral part of our life, like a TV remote control or an alarm system on a car, but until that happens, I would like to tell you about how IP telephony is implemented and what technologies are used for this in MTT.
Protocols
IP telephony is based on several standardized communication protocols. The underlying protocol is the H.323 standard from the International Telecommunication Union (ITU), according to which transnational Internet telephony operators operate. All calls in this protocol are controlled by the Gatekeeper (gatekeeper, zone controller) - a program or device that is located at the IP-telephony provider that coordinates all clients that are united in a common zone. It manages terminals, gateways, conference management devices (MCUs), etc. Thanks to the gateways, the H.323 protocol can integrate telephone and IP networks. So, the gateway can be an adapter for analog phones (eg HandyTone, GXW), which connects to the Internet and an analog phone. When dialing a number on a traditional telephone, the gateway transfers the call to the Internet, and when a call comes in via IP telephony, it allows the owner of an analog device to receive a call without using the public switched telephone network (PSTN). The gateway can be either special IP phones (manufacturers: Gigaset, Panasonic, Cisco, or Yealink), or programs on a computer to which a headset with headphones and a microphone is connected. Terminals are the end devices from which a call is made: smart phones and PCs with special software, IP phones. For a conference in which more than 3 people can participate, the MCU server is used.
The SIP protocol is considered to be more flexible and scalable, which is used by almost all IP-telephony providers, allowing not only to talk over the Internet, but also to exchange video, instant messages, play online games, etc. Incidentally, the YouMagic IP-telephony service from MTT is built on it, by the way. This protocol regulates only the procedure for establishing a connection between devices, and a network streaming data transfer protocol (SDP) is connected for data transfer. SIP has more functions than H.323, and this difference is due to the fact that this protocol was not developed by ITU telephonists, but by an open group of the international community of designers, scientists, network operators and providers (IETF). SIP protocol requires proxy servers that are installed with IP telephony providers and user terminals, that is, devices through which communication takes place. They can be the same IP phones, gateways, as well as computers or smartphones with installed software (softphone) and connected to the Internet.
There is also a corporate SCCP protocol that works according to the H.323 logic. SCCP was developed by Cisco to reduce communications costs in companies by saving on hardware.
')
Codecs
For the quality of voice transmission in all protocols, codecs are responsible - algorithms for converting voice into the required digital format. In IP-telephony, voice is converted into so-called packets, which are transmitted over Internet networks. For voice transmission in good quality, without delay, echo and distortion, turning the voice of a person into the voice of a robot, one or another codec is selected, depending on the Internet connection, program settings and other factors. For example, a single codec (for example, g722) with a fast and confident Internet connection can transmit voice in high quality, but the low bandwidth of the channel will make the conversation impossible due to the loss of phrases or sounds. In MTM's YouMagic, codecs such as iLBC or Silk can transmit voice without noticeable delays when the quality of the Internet channel is low, but they compress the data so much that some voice features are simply lost when encoding-decoding, and even at high speed they are still will not be able to provide high quality dialogue.
SIP Technology Service - YouMagic
Consumers have long been accustomed to comfortable hearing when communicating through fixed networks and even through networks of mobile operators, so any interference and failures in IP-telephony can discourage the desire to use it. Being well aware of this, each Internet telephony operator, when developing its products, has to closely monitor the status of its proxy servers and other equipment, data passing through the networks of other operators and other costly issues, while striving to keep low rates for communication with the whole world.
One of the clearest examples of maximizing the benefits of SIP is YouMagic, launched in 2011. In this purely Russian IP telephony project, agents can be smartphones, personal computers, IP / DECT phones, or regular phones, complete with a special gateway, as discussed above. For common operating systems, their own softphones have been released, allowing you to talk through YouMagic on an iPhone or Android, on a Windows or Mac OS computer.
To ensure good quality of conversation, YouMagic uses different codecs in different situations. The low quality of the Internet channel is typical for the domestic mobile Internet, therefore, when developing an application for smartphones, all the features of packet transmission in 3G networks were taken into account. So, in the YouMagic application for Android OS, automatic selection of the desired codec is provided depending on the Internet connection. And experienced users from the list of presented codecs can independently select those that will be used in certain Internet networks. This feature allows the application to automatically adjust to the properties of the Internet connection, improving the quality of voice transmission in the dialogue.
One of the most convenient types of Internet access, using IP-telephony, is Wi-Fi. Today, Wi-Fi coverage is becoming increasingly widespread, and quality is more stable. Thus, it was recently announced that in 2013 work will begin on creating a Wi-Fi network on all branches of the metro in Moscow, and ground transportation will be connected. Wi-Fi is available in cafes and restaurants in many cities of Russia. It seems that after a couple of years we will be able to stay in the “Wi-Fi cloud”, communicating with the whole world in any way convenient for us, and IP-telephony will become one of the most comfortable ways of communication.
Traditionally, communication between users of one IP-telephony operator is free of charge, and a call to a mobile or landline number in any country or city in the world is much cheaper than using a mobile or landline phone. YouMagic provides its subscribers with extremely affordable rates: Russia will have to spend less than 1 ruble per minute to communicate with any city; calls to such popular countries of the world as the USA, Germany or Israel are also profitable. And communication of all YouMagic owners among themselves, of course, remains free.
Source: https://habr.com/ru/post/154869/
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