VoIP TELEPHONY FROM INTEREC
What is VoIP?
Not long time ago, the quality of the IP telephony services was really bad, with distorted and delayed sounds.
Nowadays, the quality levels are really close to the ones of a conventional telephone, even though they would not replace the traditional telephone systems completely, they would be perfectly able to cover between a 80 and 95 percent of the use of these. The reduction in costs will be carried out by using the internet network for the sending of encoded audio in data.
Thanks to the servers GATEWAYS it is possible to make a call from a conventional telephone connected to an IP voice line to any telephone on the switched network in the world.
This is to say, that we not only can make calls to another PC or network connected to the internet, but also to any telephone or mobile phone in the world.
The progress in technology for voIP services has developed exceptionally during the last years, adding manufacturers as Cisco, Lucent, Intel, Quicknet, IBM and many other systems dedicated to supporting voIP.
BANDWIDTH
To understand the quality of the service and how the performance affect us when we have to make a planning for reduction of the telephony costs, it is necessary to understand some technical concepts.
Until now we have referred to the flow of required data ( 22kbps per channel in active voice ). This is to say that if a company install a voIP telephony service, let's say for example 10 outgoing lines, can carry out 10 simultaneous calls and required a width of 220kbps.
The reality is not quite so, now that they will never be exposed in the example, the 20 persons talking on the same time.
During a normal conversation by telephone, only during short intervals both persons talk simultaneously and therefore the maximum flow of data is a lot lower than 220kbps.
For this the actual technology provides a system known as disidentification of data in silence, which works NOT sending data when there is no sound.
Next we have to understand the concept of how this disidentification of data is carried out, as well as how the voice is converted in data.
There are two methods of doing this: with hardware or software.
CODEC'S
Regarding the audio signal in data, this goes through many steps to convert it to data and finally it goes through a filter "compressor".
These are called codec's.
A system can support one or more codec's. The codec's determine the comprehension factor, this is savings in bandwidth and the sound quality.
The codec's supported by a request in the communication: these might not be supported in the entire network. This gatekeeper from INTEREC supports the following codec's:
- g711-ulaw
- g711-Alaw
- g7231
- gsm
but for example if we make a telephone call to The United States, it can be the case that the Gateway of the routing in the other end, only work with the codec g711-ulaw. This is actually not a problem, now that the codec used for each service is automatically switched.
In this document you will find links with more information, and therefore we will not go deeper in to the economical viability or the technical aspects.
There is still one more detail that you should know about the voIP technology. On the internet, as well as in many other communications, one of the most used protocols is TCP/IP.
For voIP telephony you use a protocol called UDP. Which is the difference?
While in TCP/IP the sent packages require that the other part send us an acknowledgement of receipt, or in case of loss or expire on the network, should be send again, with UDP it is not like this.
This means that a header of pack of UDP is a lot smaller than the one of a TCP/IP, the packaging and the labels of the pack of data are smaller, this means that for the same quantity of content we will achieve bigger performances.
This is like this because using voIP telephony we cannot recover or resend a pack which was transmitted a few seconds ago, this already happened. The same thing happens when we listen to the radio or the TV, if an interference occurred, the information was lost, but it is not valid to send it out of time.
Wow, why is this?
Easy, lets talk about the latency. Don't worry, it would be to difficult to understand.
LATENCY
When we receive an e-mail or an FTP archive, which takes 100 milliseconds more or less, it is not important, when you work in voIP it is.
Therefore it is necessary to be able to implement a quality voIP service in your company, who has the lowest possible latencies.
INTEREC advise to use your own access network to avoid this, however you can use your actual ADSL or any other connection.
The access network from INTEREC offer latencies for all GATEWAYS under 100 milliseconds.
Make the test, with your actual connection you open a MSDOS window and write for example "ping www.cnn.com", this will reflect the latency with which you get to the site of CNN.
The normal latencies in the connections provided by INTEREC are normally of 30 milliseconds.
