ESrating: TELEPHONY ON VOIP

it shoots with an arrowESrating VoIP TELEPHONY

What is VoIP?

voipNot too much time ago, the quality of the services of telephony IP was really bad, with distorted and slowed down sounds that arrived at the interlocutor.
Nowadays, the quality levels are really next to those of the services of conventional telephony, although they do not get to replace the traditional systems completely, yes get perfectly to cover between 80 to a 95 percent of the use from the same. The reduction of costs is realised by means of the use of the Internet network for the codified shipment of audio in the form of data.
Thanks to servers GATEWAYS (Pasarelas) it is possible to originate the call from a connected conventional telephone to a line of voice IP, to any telephone of the dial-up network of the world.
This means, that not only we can carry out calls to another PC or connected network to Internet, but to any fixed or movable telephone in the world.
The advance in technology of last services voIP has evolved of exponential form during the two years, having incorporated manufacturers as Cisco, Lucent, Intel, Quicknet, IBM and many other systems dedicated to the support voIP.

BANDWIDTH

voipIn order to understand the quality on watch and how its yield affects to us at the time of a planning would be of reduction of telephony costs, it is necessary to handle some somewhat technical concepts.
Until now we have realised a reference to volume of required data (some 22kbps by active voice channel). This means that a company that installs a service of telephony voIP, we say for example of 10 lines of exit, it will be able to carry out 10 simultaneous calls, and would require a width of 220kbps theoretically.

The reality is not thus completely, since never they will be in the set out example, the 20 people speaking at the same time.
During a normal conversation by telephone, only in small intervals both speakers speak simultaneously, reason why the maximum volume of data in this case will be far below to 220kbps.
For it, the present technology provides with a system known as suppression of data in silence, that works not sending data if there is no sound.

Next a concept that we must understand, is how this suppression of data takes place, as well as the voice to data becomes.
We will say at first, that there are two ways to do it: by hardware and software.

CODEC

As far as the conversion of the signal of audio in data, this one undergoes several passages until turning it into data, and finally it is happened through “a compressing” filter.
They are the calls codec.
A system can support one or more codec. The codec determine bulk factor, this is saving of quality and bandwidth, of the sound.

The codec supported by an instance in the communication, can not be supported in all the network. That is to say, gatekeeper of ESrating it supports the following codec:

  • g711-ulaw
  • g711-Alaw
  • g7231
  • gsm

but for example we called by telephone to the United States, can be that the Gateway of routing in the other end, works only with the codec g711-ulaw. This in fact is not problem, since the codec used for each service is exchanged automatically.

In this document it would find links to but information, reason why we will not extend in depth in aspects of economic viability nor technical aspects.

It is still a detail that must know on the technology voIP. In Internet, as in many other communications, one of the protocols more used it is TCP/IP.
Then, for telephony voIP is used a called protocol UDP. What difference has?
Whereas in TCP/IP the sent packages require that the other end sends a receipt requested to us, or in case they are lost or they expired in the network, they are resent, in UDP this is not thus.
Consequently a head of a package UDP is much more small that one of TCP/IP, that is to say, the packing and labels of the package of data are smaller, consequently for a same amount of content we secured to major yield.

This is thus, because in telephony of voIP, it is not worth to us to recover or that resends a package to us that was transmistido does some tenth of second, that already has happened. It happens just as when we listened to the radio or the TV, if there were an interference, the information was lost, but it is not valid to send it outside time.

It goes, to that this comes?
Easy, we are going to speak to him of latency. One does not worry, will not be too hard to understand.

LATENCY

Although when we received an e-mail, or a file by FTP, that behind schedule 100 milliseconds more or less does not have major importance, when one works yes in voIP has it.
For this reason, it is necessary so that you can implement a service of voIP of quality in his company, that has the lowest latencies possible.
ESrating advises for it to use their own network of access, although you can use his present ADSL or any other connection.
The network of access in ESrating offers some to him latencies to all the GATEWAYS inferior to 100 milliseconds.

It tests, it opens in his present connection a window MSDOS and writes for example “ping www.cnn.com”, that will reflect the latency to him with which you arrive at the Cnn site.
The normal latencies in the connections provided by ESrating usually are of about 30 milliseconds.


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