VoIP Technology

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How VoIP works

How VoIP works

VoIP (Voice over Internet Protocol) routes telephone calls through a computer network and the internet. 

Rather than direct voice transfer, where sound is directly converted into an electrical signal at one end and back into sound at the other, VoIP equipment, as a connection through the internet, turns the sound into digital data which it then transmits.

This is the same as recording any sound into a computer. The sound is broken down into tiny packets of data which the computer can string together to enable coherent playback. VoIP works the same way but the information is sent via the internet.

A defined set of rules – codecs – governs how the sound is turned into data and back. The data is sent as a chunk of between 10 and 30 milliseconds of sound at a time. The codecs are capable of compensating for any data loss caused by faults in transmission.

A VoIP network has VoIP terminals connected in a similar way to an existing PC network. The technology works using either broadband or a leased line.

Calls to and from standard telephone numbers are routed through an interface between the internet and the standard telephone network. A VoIP provider supplies the interface. A caller using a standard telephone number will not detect any difference in service or quality when speaking to a VoIP business user.

VoIP will eventually replace the current phone system because it is a vast improvement in terms of efficiency, cost and flexibility. Call us on 01392 279999 and be ahead of the queue

Two ways

Two ways

VoIP allows you to make regular telephone calls through a broadband (or leased line) internet connection rather than a traditional analogue telephone line. It is a method for taking analogue audio signals and turning them into digital data that can be transmitted over the internet.

Currently, there are two main ways of using VoIP for business:

  • ATA (Analogue Telephone Adapter): The simplest way to use VoIP is with an analogue-to-digital converter, an ATA. The ATA lets you connect a standard phone to your computer or your internet connection for use with VoIP. It takes the analogue signal from your traditional phone and converts it into digital data for transmission over the internet. All you do is plug the cable from your phone that would normally go in the wall socket into the ATA and you can make VoIP calls.
  • IP phone (Internet Protocol phone): These phones look just like normal handset phones. However, instead of having the standard phone connectors, IP phones have an ethernet connector. IP phones connect directly to your router and have all that is necessary to handle VoIP calls. 

VoIP will eventually replace the current phone system because it is a vast improvement in terms of efficiency, cost and flexibility. Call us on 01392 279999 and be ahead of the queue 

Circuit switching

Circuit switching – how a traditional landline connection works

Existing telephone systems are driven by circuit switching – a reliable but inefficient method for connecting calls, in use for more than 100 years.

When a call is made between two parties, the connection is maintained for the duration of the call. Because two points are connected in both directions, the connection is called a circuit and is the basis of the Public Switched Telephone Network (PSTN).

During a 10 minute call, the circuit is continuously open between the two phones. Until about 1960, every call had to have a dedicated wire stretching from one end of the call to the other for the duration of the call. Therefore, for your London-Edinburgh call, switches between the two cities would connect pieces of copper wire all the way. The call was expensive, because, for 10 minutes, you actually owned a 400 mile long copper wire.

Calls using today's traditional telephony are more efficient and less costly. Your voice is digitised, and, together with thousands of other voices, can be combined on to a single fibre optic cable for much of the journey. These calls are transmitted at a fixed rate of 64 kilobits per second (Kbps) in each direction, for a total transmission rate of 128 Kbps. Since there are 8 kilobits (Kb) in a kilobyte (KB), this translates to a transmission of 16 KB each second the circuit is open. In a 10-minute call, the total transmission is 9,600 KB, ie, roughly equal to 10 megabytes. However, during a typical call, much of this transmitted data is wasted.

VoIP will eventually replace the current phone system because it is a vast improvement in terms of efficiency, cost and flexibility. Call us on 01392 279999 and be ahead of the queue

Packet switching

Packet switching – how a VoIP connection works

A packet-switched telephone network is today’s alternative to circuit switching. Because calls are digitised, the “dead air time” can be removed:

  • In any telephone conversation, while you are talking, the other party is listening, which means that only half of the connection is in use at any given time. Therefore, we could cut the file in half, down to about 4.7 MB, for efficiency.  
  • Additionally, there is a lot of dead air time in most conversations because, for seconds at a time, neither party is talking. If these silent intervals could be removed, the file could be even smaller. 
  • Then, instead of sending a continuous stream of bytes (both silent and noisy), the next step is to send just the packets of noisy bytes when you created them.

Data networks do not use circuit switching. Instead, data networks simply send and retrieve data as you need it. Further, instead of routing the data over a dedicated line, the data packets flow through a (seemingly) chaotic network along thousands of possible paths. This is called packet switching.

While circuit switching keeps the connection open and constant, packet switching opens a brief connection – just long enough to send a small chunk of data, called a packet, from one system to another. 

Packet switching is very efficient. It lets the network route the packets along the least congested and cheapest lines. It also frees up the two computers communicating with each other so that they can accept information from other computers as well.  

VoIP will eventually replace the current phone system because it is a vast improvement in terms of efficiency, cost and flexibility. Call us on 01392 279999 and be ahead of the queue

Codecs

Codecs – turning analogue audio into packets for VoIP transmission

A codec is the essence of VoIP. Short for compression/decompression, a codec converts an audio signal into compressed digital form for transmission and then back into an uncompressed audio signal for replay.

Most people have heard of mp3.  mp3 is the codec used by many personal music players and its function is to compress the (normally) CD quality audio which would take approximately 60 MB per track down to a file of approximately 3MB. 

Codecs accomplish the conversion by sampling the audio signal several thousand times per second. The G.729A codec has a sampling rate of 8,000 times per second and is one of the most commonly used codec in VoIP. It converts each tiny sample into digitised data and compresses it for transmission. When the 8,000 samples are reassembled, the pieces of audio missing between each sample are so small that, to the human ear, it sounds like one continuous second of audio signal. 

Codecs use advanced algorithms to help sample, sort, compress and packetise audio data. The CS-ACELP algorithm is one of the most prevalent algorithms in VoIP. Annex B is an aspect of CS-ACELP that creates the transmission rule, which basically states "if no one is talking, don't send any data”. The efficiency created by this rule is one of the greatest ways in which packet switching is superior to circuit switching. 

The codec works with the algorithm to convert and sort everything out, but it is not any good without knowing where to send the data. In VoIP, that task is handled by soft switches which know:

  • where the network's endpoint is;
  • what phone number is associated with that endpoint; and
  • the endpoint's current IP address.

VoIP will eventually replace the current phone system because it is a vast improvement in terms of efficiency, cost and flexibility.  Call us on 01392 279999 and be ahead of the queue 

Softswitches

Soft switches

The soft switch contains a database of users and phone numbers. If it does not have the information it needs, it sends the request downstream to other soft switches until it finds one that can answer the request. Once it finds the user, it locates the current IP address of the device associated with that user in a similar series of requests. It sends back all the relevant information to the IP phone, allowing the exchange of data between the two endpoints.

Soft switches work in tandem with network devices to make VoIP possible. For all these devices to work together, they must communicate in the same way.

VoIP will eventually replace the current phone system because it is a vast improvement in terms of efficiency, cost and flexibility. Call us on 01392 279999 and be ahead of the queue

Protocols

Protocols

On each end of a VoIP call you can have any combination of an analogue, IP phone acting as a user interface,  ATAs or client software working with a codec to handle the digital-to-analogue conversion, and soft switches mapping the calls. 

How do all these completely different pieces of hardware and software communicate efficiently to pull all this together? The answer is protocols.

There are several protocols currently used for VoIP. These protocols define ways in which devices like codecs connect to each other and to the network using VoIP. They also include specifications for audio codecs.

VoIP will eventually replace the current phone system because it is a vast improvement in terms of efficiency, cost and flexibility.  Call us on 01392 279999 and be ahead of the queue