The Nitty-Gritty Of VoIP

In order to use VoIP, both parties need a broadband connection. This is a high-speed Internet connection usually provided by a cable or DSL modem. Broadband modems are usually used to connect computers to the Internet, but in the case of VoIP, computers are not necessary.

The simplest form of VoIP is a computer-to-computer voice connection. All that is required for this type of connection is a computer, a headset consisting of earphones and microphone, and VoIP software. Most software packages are free and allow you to connect to any computer running the same software. 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 land-line phones — that is, phones which are not connected directly to the Internet. This type of call is usually not free but the cost is quite a bit lower than what your telephone company charges. Some VoIP services also allow you to make calls to cellular phones.

The only time that both parties need a particular VoIP software package is when they are making computer-to-computer calls. Parties receiving land-line or cellular calls do not need any extra equipment or software.

VoIP Transmissions

VoIP is based on digital data transmission. 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 an analog signal into discrete steps which are represented by numbers. The next step is to compress the audio data using a codec (enCOder/DECoder) which significantly reduces the amount of digital data while maintaining 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.

All data that is sent over the Internet is encapsulated in ‘layers’ which aid in its proper delivery. For example, web pages may use the Internet Protocol (IP) network layer to specify destination and origin addresses, the Transmission Control Protocol (TCP) transport layer to create a connection between two computers and the Hyper Text Transfer Protocol (HTTP) as an application layer to allow the Web browser to display the web page correctly.

Most VoIP uses a transport layer called User Datagram Protocol (UDP) which is faster than TCP. A commonly used application layer is Real-time Transmission Protocol (RTP) — originally developed for delivering audio and video over the Internet. RTP provides information about the sequence of the data packets so they can be reconstructed in the correct order at their destination.

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

Even though some of the packets are dropped, there is usually still enough information to make the conversation legible. The number of packets that will be dropped depends on the speed of your Internet connection in the distance between the two parties.

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