How Voice Over Internet Protocol (Voip) Works

Introduction

Voice over Internet Protocol (VoIP) is an internet technology that enables voice and multimedia communications over computer Internet Protocol (IP) addresses. The technology is also referred to as IP telephony or voice over broadband (VoBB). VoIP refers to the various communication services such as voice, fax, and SMS that are conveyed via the internet, instead of the conventional public switched telephone network (PSTN). VoIP systems use session control protocols to manage the setting up and decoding of the information at the sender and recipient positions, the decoding process allows for the transmission of voice over an IP network as digital audio. To ensure that the voice message is not distorted, a VoIP system uses a range of codecs at different stages of the transmission process, codecs also ensure security of the service (Hersent et al 2005).

Three different kinds of VoIP tools are frequently used: IP handsets, Software VoIP, and mobile VoIP, each tool has its weaknesses and benefits. The most commonly identifiable VoIP service is Skype, which falls under software VoIP.

How VoIP Works

Pictorial presentation of how VoIP works
Fig. 1: Pictorial presentation of how VoIP works

To create a VoIP call, a user must have a broadband connection, an analogue telephone adaptor (VoIP adaptor) and must be registered with the VoIP provider. The VoIP adaptor is an analog-to-digital converter and enables a person to connect a normal phone or computer for use with VoIP. It converts analog signals to digital format for transmission. Another option is to use a softphone. A softphone is a software that allows a person to use the VoIP service over the computer, Skype is an example. After installing the software, all one needs is a headset, and internet connectivity, obviously, one has to have registered with the VoIP service provider.

One major difference VoIP calls and regular phone calls is that the former uses packet switching as opposed to circuit switching as used in the latter. In circuit switching, a connection is maintained between the two persons for the whole length of the call. Since the connection occurs between two users in both ways, it is referred to as a circuit. Packet switching, on the other hand, does not create a two-way connection, when one person is speaking, the recipient can only listen. This implies that only half of the connection is used at any given time during the duration of the call. Packet switching cuts the file sent through PSTN calls in half, thus enabling a faster transmission even on slow IP networks (Davidson 2007). In contrast, a circuit switch maintains a continuous connection between the two parties, sending a continuous stream of data (including moments of silence and noise).

When a sender makes a call through a microphone or mouthpiece, an electric signal is created inside the device. These are analogue signals and cannot be transmitted over IP networks. The signal is converted immediately to a digital data using a program written on the converter. The digitized voice data is wrapped up in packets and transmitted over an IP network through the provider, where it passes through gateways and servers to the recipient. If the recipient is using a PSTN, the server creates a connection to the PSTN and directs the voice signals there. At the recipient end, codecs convert the voice from the digital to analogue format so that the recipient can receive the message. During the whole duration of the call, various protocols are used to manage the call, for example, connecting the users, dialing, and disconnecting, other protocols are used for consistent transfer of packets and to maintain the call standards.

It is important to note that the encoding and compression processes do not affect voice quality significantly. Although some components of the voice are lost, this affects only the sound signals with frequencies that cannot be perceived by the human ear. Moments of silence are also removed before the data is sent.

How Voice is Digitized in VoIP Calls

Conversion of voice signals from analogue to digital format is necessary to enable voice transmission over the internet. It can be achieved in various ways:

  • PCM (Pulse Code Modulation): this method samples the sound signals at a constant rate (8000 times/sec) and creates a number equivalent to each sample. This technique does not take into account any particular characteristic of the voice signal, hence, it can be used to digitize all types of sounds (Wittenberg 2009).
  • LPC (Liner Predictive Coding): this method takes on particular characteristics of the human voice and uses a more intricate set of procedures to convert and compress analog sounds to digital form. It functions well for transmitting human voice as it reduces the data to be transmitted, however, it is not suitable for sending music or fax due to degradation of quality.
  • SBC (Sub Band Coder): this method transmits sounds in terms of frequencies instead of sampling at constant rates.
  • Hybrid coders such as CELP (Code Excited Linear Prediction) employ a combination of the methods mentioned earlier to compress digital sound signals while maintaining voice quality (Wittenberg 2009).

Advantages of VoIP over PSTN Connections

The most commonly citeed advantage of the VoIP service is the low call costs and flexibility. Packet switching transmits less than half of the data transmitted over a similar call in traditional PSTN calls. The VoIP network runs on an existing internet network and eliminates the requirement to pay separate charges for sound and data transfer. On flexibility, VoIP calls can be made anywhere, as long as there is a stable internet connection, unlike PSTN calls that can only be made in areas where the home network can be accessed. Several VoIP calls can be made over one broadband connection, this is not possible in PSTN calls (Ganguly & Bhatnaga 2008). VoIP calls can take advantage of several web applications such as e-mail, video conversation, and video and audio conferencing to improve user experience and increase applicability. Finally, VoIP calls can be more secure over long distances than traditional phone lines.

Disadvantages of VoIP

Communication over an IP network is less reliable in comparison to PSTN networks. Due to a lack of quality of service assurances, data packets may be lost, or not be delivered in sequential order. Slow or congested IP networks may cause delays in packet transmission, causing inconveniences to users.

Conclusion

With increasing penetration of internet network all over the world, many people are set to adopt VoIP as a medium of communication rather that the traditional PSTN lines, due to the advantages attributed to this medium of communication. Communication over an IP network brings the benefits of cost and flexibility. Low costs arise from the use of packet switching, rather than circuit switching as used traditionally.

Reference List

Davidson, J. (2007). Voice over IP fundamentals, 2nd Edition. Indianapolis, IN: Cisco Press.

Ganguly, S., and Bhatnaga, S. (2008). VoIP: wireless, P2P and New Enterprise Voice Over IP. West Sussex: John Wiley & Sons.

Hersent, O., Petit, J., and Gurle, D. (2005). IP telephony: deploying voice-over-IP protocols. West Sussex: John Wiley & Sons.

Wittenberg, N. (2009). Understanding Voice over IP Technology. Delmar: Cengage Learning