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voice over IP (VoIP)

The IP protocol is suitable for data transfer, but also for voice, data and video transmissions. Therefore, voice communication and video communication via the Internet forms an interesting, cost-effective alternative to fixed networks and mobile networks. As Internet telephony, Voice over IP (VoIP), voice communication competes with classic analoguetelephony, but also with other Voice over X technologies.

Internet telephony, where voice is transmitted using the IP protocol, is about ease of use, reliability and voice quality. The latter is determined by the real-time behavior of transmitting voice packets using the IP protocol. and minimizing data packet loss and delay times. and expressed as an MOS value.

Functions of Internet telephony

Functionally, Internet telephony over a VoIP network is comparable to telephony over fixed or mobile networks. It involves the bidirectional transmission of voice signals between two communication partners. In all networks, the transmission principle consists of establishing a connection, transferring the call and terminating the connection.

In contrast to fixed networks, where a fixed connection is established between the communication partners, in Internet telephony the compressed, digitized voice signals are packed into smaller voice data packets and transmitted over the Internet. For data transport, VoIP uses the Realtime Transport Protocol (RTP) or the secure variant, the Secure Realtime Transport Protocol (SRTP). These protocols are based on the User Datagram Protocol (UDP). This does not know any confirmation, but therefore has lower delay times than the Transmission Control Protocol (TCP). Underneath the UDP protocol lies the IP protocol as the network protocol. The target station is the VoIP telephone or softphone, which is preceded by a VoIP adapter. In this adapter, the data packets are reassembled into a continuous voice data stream.

Unlike classic switching technologies, where the subscribers are identified by their phone numbers, the IP address cannot be used as a phone number in IP telephony, as this can change. Therefore, the call numbers of the subscribers are stored and updated in a server, and the connection data are queried by the server before the telephone call. In addition, the signaling used to establish the connection is done using a separate signaling protocol such as the Session Initiation Protocol (SIP) or H.323 from the International Telecommunication Union (ITU).

The voice quality of Internet telephony

The voice quality of Internet telephony is affected by the propagation del ays of voice signals, packet loss during data packet switching, jitter and noise. The propagation times and delay times depend on the conversion speed of the codecs, the signal propagation times in the switching nodes and the protocol-dependent packet repetitions. In order to keep these propagation times as short as possible, VoIP does not work with the TCP protocol, which provides for the repeated transmission of data packets with errors, but with the connectionless UDP protocol. With this transport protocol, the erroneous voice packets are not retransmitted, which reduces the signal propagation delay.

Delays in an IP network during voice transmission

Delays in an IP network during voice transmission

The voice codecs standardized by the International Telecommunication Union (ITU) for voice transmissions on the Internet have propagation times of 25 ms to 100 ms, depending on the bit rate. Depending on the buffer delay, the total runtimes can be so large that they negatively affect the voice quality. As far as voice quality degradation due to packet loss is concerned, the ITU has taken this into account in the G.114standard.

In general, Internet telephony offers significantly higher voice quality compared to ISDN or POTS. VoIP supports HD tele phony according to G.722 if this is used.

The services of Internet telephony

In addition to voice quality, the acceptance of VoIP telephony depends on the telecommunications services and features provided. This includes, among other things, the integration and use of existing telecommunication systems so that the user can continue to use his familiar telephone infrastructure with all its known features.

IP-based telephony via the Internet is designed to replace conventional private branch exchange technology and provides the basis for the integration of voice, data and video services, such as those used for web conferences, application sharing or in call centers. Further aspects for the acceptance are the global call number conversion between the IP addresses of the Internet world and the E.164 call number plan of the telephone network, the authentication of the caller and the billing possibility according to different tariffs. The RegTP has reserved the number range 032 for Internet telephony.

Due to the standardization of the operating functions for data and voice, synergy effects can be used. In addition, Internet telephony creates standardized environments with interfaces to conventional PBX environments and to public networks.

VoIP telephone Gigaset from Siemens

VoIP telephone Gigaset from Siemens

VoIP voice communication takes place primarily as phone-to-phone telephony, without a personal computer. However, there is also PC-to-phone and phone-to-PC communication. VoIP telephones, which look like classic telephones, and headsets are available as communication terminals.

VoIP is characterized by the international ITU standard for voice, data and video communication over packet-oriented networks. This standard is an extension of the H.320 standard for video conferencing over ISDN. H.323 includes data packet-switched networks in the definition. Based on the RTP protocol, H.323 can also be used for video transmissions over the Internet. In addition, the SIP protocol, MGCP protocol, MEGACO and Stream Control Transmission Protocol (SCTP) should be mentioned, which have an influence on the technology of VoIP.

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Englisch: voice over IP - VoIP
Updated at: 31.03.2018
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