The Basics of VOIP

What is VOIP? It stands for Voice over Internet Protocol, and it is a technology that allows you to make telephone calls over the Internet, rather than a regular phone line.

In order to use VOIP, both sender and receiver need to have a broadband connection. This is a high-speed Internet connection usually provided by cable or a DSL modem. Broadband modems are usually used to connect computers to the Internet, but in the case of VOIP, computers are not necessary, because they can connect telephones.

High-Speed Connection

The simplest form of VOIP is a computer-to-computer voice connection. The only requirements for this type of connection are a computer with a broadband connection, a headset consisting of earphones and microphone, and VOIP software.

Most VOIP software packages are free (you can download one from the Internet at versiontracker.com), and they allow you to connect to any computer with VOIP software. The only time that both parties need the same VOIP software package is when they are making computer-to-computer calls. There is no charge for this type of connection, and calls can be made to anywhere in the world.

VOIP software can also be used to connect to landline phones, that is, phones that are not connected directly to the Internet. This type of call is usually not free, but it costs quite a bit less than your telephone company charges. Some VOIP services also allow you to make calls to cellular phones. Parties receiving landline or cellular calls do not need any extra equipment or software.

Voice Becomes Data

VOIP is based on digital data transmission. So, the first step in any VOIP call is to convert the analog signal of the human voice into digital data. This is done within an Analog-to-Digital Converter (ADC) that divides the analog signal into discrete portions, which are represented by numbers. The next step is to compress the audio data using a codec (enCOder/DECoder), which reduces the amount of digital data as far as possible without sacrificing audio quality.

The compressed digital data can now be sent over the Internet. The data stream must be divided into packets which, besides containing the audio data, also have information concerning their destination and their place in the data stream.  

Multi-Layer Transmission

All data that is sent over the Internet is encapsulated in discrete layers, which aid in its accurate delivery. For example, a network layer specifies destination and origin addresses, a transport layer creates a connection between two computers, and an application layer might allow a page to be displayed correctly at the receiving end.

The transport layer used by most VOIP transmissions is called User Datagram Protocol (UDP), which is a very high-speed protocol. A commonly used application layer is Real-time Transmission Protocol (RTP), which provides information about the sequence of the data packets so they can be reconstructed in the correct order at their destination.  

Data Becomes Voice

RTP also has the ability to drop packets if they do not arrive within a certain time limit. This is necessary for telephone conversations because if the telephone software waited for every packet to arrive before reassembling the voice, it would cause unacceptable delays in the audio stream.

Even though some of the packets are dropped, there is usually enough information left to make the conversation understandable. The number of packets that are dropped depends on the speed of your Internet connection and the distance between the two parties.

Once the voice data has arrived at its destination, it is reassembled in the correct order and converted from digital back to analog -- the sound of a voice.