Voice on network IP

The voice on network IP , or “VoIP” for Voice over IP , is a technique which makes it possible to communicate by the Voix via Internet or very other network accepting the protocol TCP/IP. This technology is in particular used to support the service of telephony IP (“ToIP” for Telephony over Internet Protocol ).

Description of operation

See CORBA (Common Object Request Broker Structures)

Principal protocols

The principal protocols used for the establishment of connections in voice on IP are:
  • H.323 ;
  • IAX (Asterisk);
  • SIP
  • MGCP ;
  • UNISTIM (owner Nortel)
  • SCCP (owner Cisco Systems);
  • UA/NOE (Alcatel owner)
  • Jingle, based on the protocol of instant messaging open standard Jabber.

The principal protocols used for the transport of the voice itself are:

  • RTP;
  • RTCP .

Various modes of diffusion

The term “VoIP” is in general used to describe “point-to-point” communications. For the diffusion of sound on Multipoint IP in S, one will speak rather about Streaming, like the radios Web for example.

The voice or the sound on IP can be made in mode Unicast, broadcast or multicast on the networks, i.e. in “point-to-point” mode, mode “an emission and several receptions” (like a transmitter TV, e.g.) and in mode “an emission for several receptions” (but the signal is not road that if there are receivers). The protocol H.323 functions only in Unicast mode.

The transport of communication on IP is very depend on the Délai of latency of a network. This time influences much the psycho-acoustic quality of a conversation. With the advent of the networks 100 Megabit s/s and ADSL, the latency times become acceptable for a daily use of the voice on IP. Contrary, connections by satellite connection suffer from a latency time often too important to deal with the applications of voice on IP. On average, the latency time on this type of connection is estimated between 400 and 800 milliseconds. A telegraphic connection (Fiberoptic or Copper) profits from a latency time from 60 to 200 milliseconds. More than latency, it is the gigue ( jitter ) which penalizes the voice on IP. Indeed, if there are fluctuations of the signal in amplitude and frequency it is necessary a mechanism of resequencing of the packages in order to restore the text message, process which will result in white and waitings.

Software aspect

With the vulgarizing of the high speed networks the number of possible applications increased considerably. The applications of VoIP ( Voice over IP ) are one of the new possibilities offered. Indeed, the increase in the flows and connections permanent offer possibilities of development of the voice on IP (Internet Protocol).

The development of VoIP trained the originators of platforms of programming to develop API ( Application Programming Interface ) specific to the voice on IP. The integration of new needs in a platform for development makes it possible to attract the originators of software which must integrate functions of voice on IP in their applications. They implement the protocol SIP.

The API of VoIP can be used in many applications, simplest being the software telephones ( software phons ). Other applications can integrate of VoIP like secondary need. Let us quote for example the applications of Instant messaging which more and more often integrate the possibility of speaking directly with its contacts or all the applications requiring a textual interaction between the various customer applications like the video games.

Some voice software on IP

Software owners of suppliers of switch

The large majority of the manufacturers of traditional telephone switches produce and market solutions of VoIP based on their own technologies:

Other software owners

Free software (open source)

Material

One finds more and more material directly compatible with software of VoIP like telephones Wi-Fi. The majority are bound, in their operation, with the solutions owners like the telephones Skype . The professional solutions are related as for them to the open protocol SIP; the terminals are in this case compatible with the corresponding free software. There exist interface boards NCV which make it possible the softPBX (such as Asterisk) to be connected to one or more phone lines.

IPv6

The transfer of the voice on IP should be improved by the protocol IPv6 when it is more widespread. The Quality of service integrated in its design makes it possible to manage priority flows such as the voice.

See too

Internal bonds

External bonds

  • French-speaking VoIP gate
  • voip-info.org: wiki anglophone dedicated to the voice on IP

References

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