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How VoIP Works – A Non-Technical Discussion
Introduction
For anyone considering trying VoIP telephone service, a basic understanding of how it works could be very helpful. This article will attempt to explain, in non-technical language, how VoIP technology works. We will give a definition of VoIP, and then explain the various components by describing what takes place from start to finish during a VoIP telephone call.
Definition of VoIP
VoIP stands for Voice Over Internet Protocol. It is actually a set of protocols, or standards, that have been developed over a number of years. These protocols allow a voice conversation to take place over the internet between two or more callers. Special protocols are needed to do this because a voice-conversation, unlike regular internet traffic, requires that a continuous data stream be transmitted without interruption.
The first VoIP protocols were developed in 1996 by a company called VocalTec. VoIP has since that time evolved into a group of protocols. Two of the most commonly used are referred to as H.323 and SIP. H.323 was initially the most widely used, but SIP has recently become the more widely used protocol.
In addition to the VoIP protocols, specialized software and hardware are required to utilize VoIP. We will discuss this software and hardware by tracing an internet telephone call from end-to-end across the internet, briefly explaining the components necessary at each stage of the call.
A Practical Example
For the purposes of this example, let us assume that you are placing a call from Los Angeles using an ATA (Analog Telephone Adapter) with a regular analog telephone connected to it, and that call is being made to someone in New York who has only regular telephone service. This will allow us to discuss most of the components being used in VoIP today.
When you pick up the handset on your analog telephone, you will hear a dial-tone, an analog signal, which is provided by the ATA. Next, when you dial the phone number, the ATA will convert the analog touch-tones into a digital number that is sent via your high-speed internet connection to your VoIP provider’s server on the internet. The VoIP server will determine the location you are calling, and route the digital call data to another server in the New York area. This server will have an interface to the PSTN (Public Switched Telephone Network – the traditional local phone company) in the New York area. At this point, the digital data will be converted back into analog signals, allowing the analog user your are calling to hear their phone ringing and answer the call.
Now a “connection” has been established between your VoIP phone and the analog phone belonging to the person you are calling. When the person in New York answers the call, their analog voice signals will be transmitted via the PSTN to the local VoIP server. At this point, the sound of their voice will be converted into digital signals and routed to the main VoIP server on the internet.
This server will in turn route the data back to your ATA, based on its IP address. Your ATA will then convert the digital data into analog signals that are fed to your analog telephone handset. The result will be your hearing the sound of their voice. At this point, a simultaneous two-way conversation can be held until the call is over.
As you can see, VoIP is not a simple technology. However, the hardware and software components required to use VoIP have advanced to the point that placing a VoIP call “appears” the same to the user as placing a regular telephone call. It is this ease of use, plus the cost savings and enhanced features of VoIP that are contributing to its vigorous growth.
Next, we will look at some of the specific advantages, including lower cost and enhanced features, provided by VoIP service.