DATA VoIP SOLUTIONS: THE FUTURE TO COME

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Where Am I > Data-VoIP-Solutions/How VoIP Works
 
 
 

How VoIP Works

How VoIP Works

 

In modern conventional phone systems, your speech is converted to digital form so that it can be transmitted in binary through fiber-optic cables. These are transmitted at a speed of 8 kilobytes per second (8 KBps). Since this a two way signal, the transmission rate is a total of 16 KBps.  Much of the connection time during a telephone conversation is silent, and during the talking only one party is normally speaking while the other listens.  The connection time is then only working at 50% efficiency at best.  

 

It is obvious, therefore, that transmission speed could be more than doubled if the silent time could be removed and the two way connection used more efficiently. nbsp; If we work out the file size for a normal conversation of, say, 15 minutes we find that it is:
 

  • 16 KB per second
  • 960 Kilobytes per minute
  • or 14,400 Kilobytes for the 15 minutes.
That is 14.4 Megabytes (MB).

If this was sent out as a single file, it would require a 14.4 MB file to be transmitted.

If we could take out the silent time and combine the digital signal of both speakers so that the two sets of conversation are combined, we could reduce this file size to around 7MB.This could be transmitted at less than half the time of the 14.4MB file. This is the basis of what is known as ‘Packet Switching’.

PACKET SWITCHING

In packet switching, small packets of information are transmitted as they are generated, rather than the data streamed when generated as in ‘circuit switching’. The way this works is:

  1. The computer you are using for your VoIP connection cuts the data into small parcels, or ‘packets’.
  2. It then addresses each packet with the IP address of the receiving computer.
  3. Each packet contains what is known as a ‘payload’, which is a small piece of the file being transmitted.
  4. The computer then sends the packet to a ‘router’, after which it is finished with it and starts work on the next packet.
  5. The router sends the packet in the direction of the receiving computer to another router, and so on until the packet reaches the receiving computer.
  6. This continues till the receiving computer has the packets, and reassembles them into the original file using a set of instructions with each packet, much as you get instructions for assembling flatpacks.

The packets take the shortest route to the receiving computer, so,  depending upon the traffic on the internet at any one moment, each could take different routes to eventually reach their destination.

This type of switching is extremely efficient: the packets are much smaller than the original whole and take the fastest route they can find along the system of routers. It bypasses congestion very effectively, and using this technology, VoIP calls can take up to a quarter of the time required for the same length of call using conventional circuit switched telephony.

Further reductions in file size, and transmission time, can be made by data compression in much the same way that MP3 compresses music files. In modern VoIP systems, codecs are used to achieve this.

DISADVANTAGES

The first, and possibly a major, disadvantage is that by its very nature, VoIP cannot rapidly be traced to a specific geographic location, so emergency calls (911, 999, 211) cannot be traced to specific locations if the caller fails to, or is unable to, provide an address. Technology would have to improve for this to be possible.

VoIP, by the very fact that it relies on internet connections, normally relies on electrical power. Unlike conventional phone systems, if your computer is not powered up, you can neither receive nor make the call. The VoIP system is therefore dependant on a continuous electrical supply; unlike cell phones or circuit switched phones unless you have a wireless multi-handset system working from a powered source, VoIP cannot replace standard phone lines for such services as digital subscription TV and digital VCRs. Where these services are in use, not to forget connected security services, the telephone line must be maintained.

All packet switching systems are susceptible to the packets being placed in the wrong order, or some packets failing to arrive. When packets arrive late, due to taking a longer route along the internet, it is called latency.

These, and other associated problems, are what could render VoIP unattractive to the corporate user until they have been resolved.

These possible deficiencies can result in phone conversations becoming distorted or even lost, though the problems are currently being addressed and resolved. So, assuming that they are, how are the VoIP files transmitted and received in an efficient enough manner to render the technique attractive to the customer, the end-user?

The answer is by means of codecs, soft switches and protocols. Most people familiar with compressed audio and video files, such as MP3, and MPEG-4 know about codecs. Codecs take a signal and convert it to a compressed digital format. This reduces the file size and allows it to be transmitted quickly, and re-constituted by the codec in the receiving computer to the uncompressed format for replay.VoIP could not exist without codecs. They are used in every form of audio and video file transmission, including not only music and movies, but also video games.

A codec, which stands for coder-decoder, or compressor-decompressor converts an audio signal into a compressed digital signal for transmission and then back into an uncompressed audio signal for replay. There therefore must be a codec at the transmission end and the same codec at the receiving end. This is the essence of VoIP: the compression of data voice signals and conversion back to the original speech in an identifiable and intelligible form.

The next articles to posted here will explain what soft switches and protocols are and how they help to ensure that VoIP is the telephone system of the future.