Telecommunications Industry: The Advent of VoIP


The paper described here is a theoretical research paper on Voice over Internet Protocol (VoIP). “VoIP is a revolutionary technology that has the potential to completely rework the world’s phone systems.” (Valdes & Roos, 2010, para.3). It can be defined as a technology that enables Internet users to communicate with others using the Internet. It is also popularly known as IP telephony, Internet telephony, and voice-over broadband. It is very easy for being a VoIP user; just log in to the VoIP account at the VoI providers and use it. Most of the providers include services such as caller ID and call waiting. There is much software available in the market to assist Internet users to communicate. Some of them are Skype, MorphVOX, ZFone, and SkyQube, which can be readily downloaded with ease. “This phenomenon has made a profound change in the world of telephone communications. The traditional method of making calls to the landlines is being fats replaced by this technology that has taken the world by storm. Not only is this method economical as this does not involve the telephony company charges that are pretty heavy, it also gives you better coverage.” (VoIP history, 2008, para.1).

This paper as a whole follows qualitative research, which is a distinctive method of theoretical research that develops a clear vision on the basis of ideas from observation, case studies, records and talks, describes the theories, principles and results. This paper aims at looking at the observation and studies of the phenomenon.

The thesis of this research paper is that, even though the idea regarding Voice over Internet Protocol emerged very long years ago, it did not come into existence because of many reasons. Changes took place during that time itself and everything happened just like what Moore law stated. The upcoming VoIP has bought a significant change in the telecom industry by adding lots of extra features to the existing telecom scenario to enjoy more benefits and also this has got the ability to be wirelessly wired at all times regardless of the location. The main intention of this paper is to explore different scenarios of this technology from its past to present.

Research question

What sort of revolution bought a significant change in the telecom sector or what paves the way for a change in the current or existing telephone sector?


The introduction of new technology like VoIP had bought a great sort of revolution in the telecom sector.

Limitation of PSTN

One of the main limitations of the Public Switched Telephone Network is its high cost for communication. The portability of the PSTN is another limitation.

Merits and demerits of VOIP over PSTN

One of the main merits of VOIP over PSTN is its low cost. It is possible for a user to make a pc-to-pc call free of charge but it costs less for a pc-to-phone call and is much cheaper when compared to local phone calls.

The VoIP calls are also portable. The VoIP services can be used from anywhere if there is a broadband connection. Signing in into the VOIP account enables it and it can be used while traveling too, keeping users in touch with their clients, families or business associates round the clock. Since each of the users has specific user ids and passwords, it’s possible to use them irrespective of time and place safely.

Other features that are helpful for the users are automatic redialing call waiting, call forwarding, voice mail, caller id and phone conferencing.

Demerits include; power is a major requirement for the VOIP telephony. Continuous power service is important, as the current cannot be enabled through the telephone wire, to power the telephone.

Thereby, when the power is off the phone is dead. It will be unable to trace the position of the VoIP user by emergency calls like 911 but could be traced by the traditional phones.

VoIP sound gets scrambled as they pass through the Internet, thus bringing a problem in the sound quality and reliability.


VoIP is found to be one of the fast-growing computer technologies in this present age. It is the technology that enables us to make telephone calls using a broadband Internet connection. VoIP is one of today’s most explained topics that have immense popularity around the globe. The growth of the VoIP didn’t happen all of a sudden; it took years to be what it is now. The first development, that came into being in 1974 for the VoIP was done by the Institute of Electrical and Electronics Engineers, by publishing a report on Protocol used in ‘Packet networking’. This paper revealed issues on the existing protocols for packet switching networks that were used during that time. The paper insisted on a single protocol called TCP, as there was more than one network. This greatly influenced the development of the Internet.

The Internet protocol IPv4 was introduced in the year 1981. After 4 years the National Science Foundation commissioned the NSFNET which was a network connecting the computer departments of the universities.

But it took 10 years for the next development in 1995 for the release of the first Internet phone software. Though it faced many issues like sound quality, delay in connectivity, etc, they were able to make free calls.

The actual developments in Internet telephony started with the introduction of commercial Internet software by VOCALTEC in 1995. Though the software was designed for a home pc, it supported devices like headsets, microphones, soundcards, and speakers. A new protocol was introduced in software, which was then very well accepted in markets. Even though the cost was very low it lacked voice quality and the availability of broadband. Although there were many inabilities made by the software, it proved to be a great advancement in technology which was remarkable. With this, the technology saw a significant improvement in the telephone network. Many more companies came into the race providing many more offers. Other such providers were Lucent and Cisco. By the year 2005, most of the issues were cleared out like voice quality and broken calls.

The revenue assured by the introduction of VoIP tends to increase exponentially. This can be due to its wide use and low cost and services provided. The introduction of videophones has fuelled this situation to a higher level.

Commercial Markets and Applications

Why is VoIP being adopted so readily in applications of business, distributed sites and remote/ home workers?

Today VoIP plays an important role in business applications as it provides great savings to them. Like the traditional phone services the VoIP permits long assured videophone abilities at cheap rates. Therefore it helps to save the business expenditure, progress the consumer services and so on. Through the VoIP lines, any phone can attain its benefits because there are many Internet connections utilizing the VoIP technology so that it facilitates international telephone networks. This technology is very useful for businesses if they have many branches in different parts of the world as they can make calls at a reduced cost. And also it can carry out the business activities in an efficient manner by using this technology. Moreover, the business with distributed sites can reduce the phone cost by connecting to the same telephone network.

In the business application, the VoIP helps in congregated conferencing as itAllow your employees to connect with remote customers and business partners for presentations, proposals and live product demonstrations. Audio conferencing, document sharing, application sharing, enterprise instant messaging, the ability to schedule one-time, recurring, or reservation-less calls and integrated audio and visual recording are supported in this easy-to-use, cost-effective collaboration tool.” (Advantages of VoIP, 2010, para.11). In an organization, the staff can communicate easily and rapidly through this incorporated communication infrastructure. The remote/home workers also benefit through the use of a VoIP system. Today many people are working from home therefore using VoIP technology they can successfully work from home. It facilitates to reduce their traveling cost, effort, etc. The home workers can do their work in the same way as doing it from the office. VoIP provides more flexibility to the phone system. The Internet-based phone offers the same functionality when it takes from the organization to the home office. With the advantage of clarity and quality of VoIP, the demand for this technology is increasing day-by-day in the business areas.

VoIP Systems Architecture

Voice over IP (VoIP) is a technology used to transmit voice as well as video signals through the internet using the IP. Because of the ability of VoIP to transfer video with the voice technologies like PBX and PSTN are becoming rarer and rarer in offices as well as in homes. Therefore it is a great challenge for the providers to maintain the existing demand for VoIP among the users. There are three main VoIP architectures developed by the standard bodies with the help of different protocols. They are H.323, SIP, and MGCP/MEGACO/H.248. The architecture of VoIP is shown below.

VoIP architecture and technologies.
(VoIP architecture and technologies, 2010).

Working of VoIP

The working of VoIP can be compared to that of the working of the microphone in recording a sound. In a microphone, the human voice is stored as simple samples of sound which is actually a tiny bit of human voice at a high bit rate. This is actually stored in the user’s computer memory itself. This is the difference when it comes to VoIP where the voice is stored not in the local computer but in the other computer with whom they are communicating. One of the main differences is that the voice samples are broken down into smaller bits so that they are trouble-free in transferring through the wire. The computer can collect all of these samples and play them at once, thereby enabling the listener to hear them with ease.

This is the inside part of ‘how VOIP works?’. The voice that is sampled first gets compressed thereby taking the lowest space and then it is transferred. The computer uses many types of CODECS to compress/decompress the voice samples which reduces the bandwidth. CODECS also improves spoken sounds though it is not as clear as spoken directly which has a greater effect on human grounds. These data are collected as a whole and transmitted over the IP network (Packetization). Most probably it is at the rate of 10 packets per 20 milliseconds.

Though the VoIP is cost-efficient and fast, it has a tendency to lose data. Maybe it can be a single sample of data. It can be mainly because low bit taking capacity of the network or the low input rate. Some packets even get delayed to be sent to the dedicated party itself. Some of these packets which are delayed for a longer time are discarded. Change in delay in sending data is called jitter. A good VoIP is one that maintains sent/sent back delay at correct intervals.

(JPG image, n.d.).

VoIP Protocols

The execution of the Voice over Internet Protocol facilitates transmission of the voice traffic in an Internet Protocol network. In order to offer effective VoIP communication services, many protocols are utilized in this process like SIP, H.323, SGCP, MGCP and so on.


The Session Initiation Protocol (SIP) is an indication protocol, which is utilized for managing communication sessions in an Internet protocol. “A session could be a simple two-way telephone call or it could be a collaborative multi-media conference session. The ability to establish these sessions means that a host of innovative services become possible, such as voice-enriched e-commerce, web page click-to-dial, Instant Messaging with buddy lists, and IP Centrex services.” (What is SIP introduction, 2010, para.1). For the purpose of signaling, the Voice over Internet Protocol society had implemented the Session Initiation Protocol. This protocol is called the request-response protocol, which is similar to the SMTP and HTTP protocols. As a result, SIP takes place beside the applications of the internet. With the usage of this protocol, the telephony incorporates the other services of the Internet without any effort moreover it develops into other internet functions. For the joined multimedia and voice services the SIP is an excellent and simple technique.


“Most voice over IP (VoIP) applications utilize H.323.” (Mitchell, 2010, para.2). This protocol is used for multimedia communications, which carry out instantaneous transport of video and audio data over the Internet protocol. The important aspect of this protocol is Quality of Service, which permits the management of traffic in the process of packet delivery. Moreover, it also improves voice quality. H.323 includes various aspects to deal with the breakdown of intermediate network elements. For the appropriate broadband and narrowband connection, this protocol encodes the messages in a compressed binary layout.


In order to manage the telephony gateways from exterior call control components, the Media Gateway Control Protocol (MGCP) is employed. This protocol presumes the architecture of call control. “MGCP is a protocol for controlling media gateways from call agents. In a VoIP system, MGCP can be used with SIP or H.323. SIP or H.323 will provide the call control functionality and MGCP can be used to manage media establishment in media gateways.” (Media gateway control protocol, n.d.). The MGCP is utilized among the media gateway and call agents furthermore it also works together with H. 323 and SIP.


“Simple Gateway Control Protocol (SGCP) is used to control telephony gateways from external call control elements. A telephony gateway is a network element that provides conversion between the audio signals carried on telephone circuits and data packets carried over the Internet or over other packet networks.” (Voice over IP: Skinny, n.d.). By focusing on the programming troubles in the call agent this protocol maintains the gateway easily. Moreover, by restricting their functions to the precise cause of gateway management, it maintains the protocol easily, and selecting the text encoding the SGCP sustains the execution simply.

VoIP Compression Methods

In VoIP, the main compression used is voice compression. Many features like bandwidth, software, hardware, etc. influence the voice quality which can be controlled and monitored. In VoIP, data compression is an important part. In this process, the compression of the voice data is done and it is made in a compressed form so that the transfer of data becomes easy. In the compression process, the voice signals are encoded into digital data which is compressed into smaller units and is carried through the internet. When these packets reach the destination, the packets are decompressed to their normal form and then translated to analog voice. This enables the user to hear the voice. The codecs are used for encoding as well as compression. The conversion of analog data to digital data is carried over IP networks. “The quality and efficiency of the compression software, therefore, has a big impact on the voice quality of VoIP conversations. There are good compression technologies and there are fewer good ones. Each compression technology is designed for specific use under specific circumstances. After compression, some compression technologies incur some loss in terms of data bits and even packets.” (Unuth, 2010, para.4). Data compression can make the VoIP’s better than the land phones considering the quality of voices. For this, the other features like software, hardware, bandwidth, etc. should be favorable. Better outputs can be obtained by lessening the load of data. Lousy compression results in the loss of some of the audio stream because of the encoding and compression of the voice data.

VoIP and QoS

QoS refers to the quality of service. The QoS on the IP networks is a topic that obtained good attention in the modern time and this has led to the development of QoS in Voice over Internet Protocol. The success of VoIP depends on QoS to some extent. It has become more complicated over the years. QoS is available for networks ranging from small LANs to large networks. Quality is an important part of networks. In the Voice over Internet Protocol, the quality is determined by the capability in listening and speaking in a clear and continuous voice without any unwanted noises. “It is very important for the Voice SP or an Enterprise to implement QoS for VoIP traffic in their networks. There are a number of factors that can affect the quality of VoIP traffic as perceived by the end-user. Some of the common factors include delay, jitter and packet loss.” (Ahmed, Madani & Siddiqui, 2009, para.2).

Service in VoIP includes the communication facilities given to the customers. The achievement of good voice quality with restricted and shared bandwidth is acquired by the quality of service. The settings of QoS can be changed to achieve high quality. The QoS is usually positioned at the level of the router. For implementing the QoS practices in the network, a router with QoS software is used to organize the quality of service needed. The quality of service configuration usually supports voice than other types of data. There would be many securities and QoS issues for an end-to-end VoIP. The QoS can be provided to the VoIP using methods like integrated services and differentiated services. “The integrated services, or Intserv, method of providing quality of service is to use a protocol for explicitly reserving bandwidth on a per flow basis. The Diffserv approach to providing QoS support differs fundamentally from Intserv in that it does not refer to a specific protocol for providing quality of service but rather an architectural framework designed to facilitate QoS.” (Gallon, 2003, pp-7-8).

QoS enables the network for providing good services to different network traffics. The main goal of the quality of service is to offer precedence including the bandwidth, jitter, latency, etc. giving priority to anyone flow which does not result in the failure of others. QoS helps to offer good services to some flows. For the performance at a required level, the network monitoring system must be installed as an element of the Quality of Service. For the new generation of internet applications like VoIP, QoS is very significant. Quality of service helps in providing precedence to the packets of the VoIP and can be carried in front of heavy packets of data in the queue. It functions well when a comparatively lesser part of the load in the network is made up of voice.

Network Delay considerations

Network delay is a significant part of computer networks. The traffic in the network has many issues that have to be considered. If these issues are not considered the network may not function properly and may result in failure. The delay is the time taken by the packet to move from one end to another end in a network. It is otherwise known as latency. Latency “is the amount of time it takes a packet of data to move across a network connection. When a packet is being sent, there is “latent” time, when the computer that sent the packet waits for confirmation that the packet has been received. Latency and bandwidth are the two factors that determine your network connection speed.” (Latency, 2010, para.1).

The large value of latency does not reduce the quality of sound in the call, but there can be a deficiency of synchronization during the speech. Usually, the latency should be smaller than 150ms. The network delay is the total of all the packet or signal delays in a multi-service network. Latency is created by the time taken to produce the voice service packets. The delays are produced by the time it needs to fill the data in the packets. The time needed to fill the packets increases with the increase in the size of the packets. If the size of the packets is less, the delay will also be less based on the software and hardware installations. Latency is also caused due to the serialization of digital data to the physical connections. If the latency is smaller, the media is also smaller. The network delay is an inevitable part of VoIP.

VoIP Devices and Systems

Nowadays different kinds of VoIP devices and systems are obtainable in the marketplace. As these devices are useful to obtain excellent voice through the Internet call, there is high demand for them in the market. With the usage of various hardware and software, VoIP devices transmit voice signals over the Internet. The voice signals are decoded by the VoIP device gateway or by another VoIP device party at the other end. Therefore, the individual whom we are calling is able to recognize clearly what we are speaking.

Analog Telephone Adapter (ATA)

One of the main general devices is Analog Telephone Adapter (ATA), which makes it feasible to employ over the internet by translating the analog audio signal from the telephone. Translating into digital data effectively performs this process. There are various types of Analog Telephone Adapters in different models and dissimilar styles. This device provides high-speed access to the internet.

Internet Protocol

The other kind of VoIP device is Internet Protocol. “Instead of the RJ-11 in home phones, an IP has an Ethernet Connection RJ-45 and connects to one’s computer’s router, and gives you that opportunity of making a direct phone call from your computer at home as you dial normally like a home phone.” (The basics of VOIP devices, 2008, para.5). This device is very simple to use.


While executing the VoIP system one of the essential elements is the VoIP gateway. The function of this device is the processing of sent and received voice signals. This gateway may be either digital or analog tools. In the case of outgoing voice signals, the gateway translates the voice signal into IP packets that are obtained from the PBX in order to send it out over the network. In the same way in the case of the incoming voice signal, the gateway brings together the data packets of IP in the manner which can be employed by the PBX.

Application servers

The application server is intended to run particular applications. This server is employed to carry out one application and is utilized for the functioning of certain types of applications. This technology is widely used in the organization as it facilitates to increase, manage and carrying out the custom applications in a secured and consistent manner.

Proxy servers

This server is a mediator between the client and server. There are various servers and when the client requests the internet resources and services, this proxy server provides the resources and services to the client by connecting to the appropriate server. Thus, the client gets the relevant resources from the internet with the help of a proxy server. In certain cases, the proxy server provides the request of the user without contacting the particular server.

USB phones

For PC-to-Phone users, USB phones are widely used due to their low price and also it is simple to use. The USB phone consists of a microphone, keypad and speaker and through a USB port, it interfaces with the computer system. In order to use the USB phone, the computer is required to be on. Another kind of VoIP device is the IP phone which is similar to the usual phone and its function is also somewhat the same as the ordinary phone. The Internet Protocol phone translates the voice signal straight into a digital signal. This phone charges only less cost when compared to the traditional phone and also offers an excellent quality of call as well as rapid accessibility of the internet.

VoIP Implementation basics

The main goals for implementing VoIP are to attain (a) significant savings in network maintenance and operations cost and (b) rapid rollout of new services. The objective is to utilize open, flexible, and distributed implementation of PSTN-type services using IP-based signaling, routing, protocol, and interface technologies.” (Khasnabish, 2003, p.1).

Stages of VoIP implementation

VoIP requires some schedule before starting the process. Among these, planning is the foremost step. Here one needs to consider very clearly what is the need and what is going to be done to meet the need. Once these are formalized than the next step is installing, running and integration. There needs better idea regarding everything in order to meet the objective. Only then it is possible to provide a better quality of services to the users. The existing network must always be ready during the planning stage. Knowledge regarding the network is very important. This always helps to assess any kind of negative effects that are happening to the applications which are currently running on the network. Also, it will help to evaluate performance issues that are happening with the VoIP traffic after the installation or up-gradation. Also, it is necessary to be always prepared to solve any issues that upraise with this. The next step of implementation is the evaluation and procurement of equipment, software and services. If the products are from different vendors it is very much necessary that everything should be run through consistent and repeatable tests in order to make sure that everything is proper and it all should match the metrics that have been specified by the vendor. Testing is also very much needed in order to see each of the vendor’s products will incorporate into the network without any problems.

Deployment and verifications are other important processes that are needed to consider as a significant factor while implementing VoIP. Deploying multimedia applications are always a great challenge. The problem can be identified only when they are in use. So, it is a prerequisite that one needs to run the network and should test the reliability of telephone calls. The uptime and reliability may take in different components. They include the following:

  • “The data network equipment along the path between the parties in a conversation, to include routers, switches, network interface cards (NICs), and cabling.
  • The complete range of telephony components includes VoIP servers and their hardware and software.
  • Whatever users come in contact with, to include IP phones, desktop computers, and their unique software and configurations.” (Ransome & Rittinghouse, 2005, p.152).

For most enterprises, the performance of a network has got an important role in the success of the business. For business purposes, the use of VoIP has become a critical factor. Service level agreements are used to measure the performance of VoIP and whether it is meeting the expectation of the users or not. Good network management is required to report what’s happening across the network and with the components. This is very important in the case of VoIP implementation.

Steps for implementation

How to implement VoIP?

Two ways are there to implement VoIP.

  1. “To buy VoIP service from a service provider and
  2. To build VoIP network on our own.” (VoIP: Voice over internet protocol architecture and features, 2010, p.121).

How to set up VoIP for the service provider?

In order to set up a VoIP for the service provider, it is necessary to choose the service provider first. Equipment is provided for the customer either free of cost or some cost. They will offer different packages for calling. Skype, Lingo, etc are different providers. Initially, signup is required with any of the service providers that are providing voice-over IP. Once it is finished one will “receive a voice over IP (VoIP) phone adapter. Configure and connect the voice over IP phone adapter to the DSL and phone.” (VoIP: Voice over internet protocol architecture and features, 2010, p.121).

How to set up VoIP over the network?

In order to set up VoIP over the network, hardware and software are needed. For example, voice-over IP gateway and gatekeeper hardware are required for Cisco VoIP architecture. Apart from this routers/switches and servers are needed to facilitate Cisco call managers for handling VoIP PBX functions.

Qualitative research for implementing VoIP

This segment of the research paper includes concise observation and a study regarding how Gunadarma University implemented VoIP in their organization. Before implementing VoIP, the 9 campus locations were using old analog PABX and communications were taking place with the help of PSTN or mobile phones. One of the disadvantages of using PSTN is it requires expensive hardware and maintenance of the system requires lots of hard work. “Mixing the computer and communication network technology become a single network comes up as the solution for business. This is called as convergence.” (Wicaksana, n.d., p.86). The basic functions of a computer start changing when people start to implement some additional features in computers rather than data sending and receiving. This bought the VoIP into the prospect. Cost reduction is very effective because VoIP is inexpensive. It can provide more services of phone system and also it is more flexible and mobile. Besides these advantages, there are some disadvantages and it is mainly regarding reliability and call quality problems. Another main concern is “VOIP services go offline if there is no power” (Patullo, 2010, para.2) and also when the internet link is down. While implementing VoIP in an organization there is no need to change the network infrastructure and there are several scenarios in order to do this.

Methodology and analysis

Approach to Implementing VoIP: There is some consideration that should be kept in mind while choosing an approach to implement VoIP. The approach mainly depends on technological expertise, budget, risk and all. Once all these are clear one can move forward.

Implementation of VoIP

Communication technology used in the Gunadarma is very costly and it does not support any mobility. In order to overcome this situation, Gunadarama university thinks to switch over their communication to VoIP. Main factors Gunadarma University considered before implementation includes operational and implementation risk, cost and budget regarding implementation. Since cost and risk are main factors it is concluded that it is ideal to implement an open-source VoIP system with the existing network. By this cost and risk failure can be avoided. Design of VoIP in Gunadarma University: Open source VoIP chosen to implement is Asterisk. The reason for choosing this is its features are very flexible, the cost is low, compatible with ample variety of platforms, supports provided are free and abundant.

Once Asterisk is chosen as a suitable VoIP system next issue is in designing a VoIP network. For accomplishing this, there are two steps that need to be considered. One is network assessment and another one is the existing phone network. A network assessment is done in order to make sure that the existing network supports the new installation. It includes checking router server, link utilization and quality of service. The link utilization by each campus has been shown below.

(Wicaksana, n.d., p.88).

Set up of phone configuration before implementing Asterisk has been shown in the below given diagram.

General phone configuration in each campus before implementing Asterisk
(Wicaksana, n.d., p.89).
Detail phone configuration in each campus before implementing Asterisk
(Wicaksana, n.d., p.89).

The analog PABX can only provide normal telephone services and in this data, infrastructure was not in build into it. Each of the campuses has been connected with wireless facilities that can handle a capacity of about 10Mbps. So by referring to the above table it has been understood that bandwidth utilization on campus was below I Mbps. So, it is concluded that there is no problem in supporting the Asterisk VoIP system.

Implementing of design

The new design redefines the existing diagram by incorporating new VoIP gear. The Asterisk can provide all sorts of standard telephone services as well as this can be integrated into a messaging infrastructure. This will enhance the users in using VoIP applications. Also, the prioritization has been done in voice traffic according to the data traffic. Voice trunks continue to be connected with the network, which is connected to a public switched network, and now Asterisk PABX is acting as a gateway to the carrier. VoIP endpoints finally joined the LAN.

Below depicts the diagram when implementing Asterisk into the network.

Implementing of design
(Wicaksana, n.d., p.92).

Gunadarma gets noteworthy benefits mainly in cost saving by implementing VoIP which is especially open source based on VoIP like Asterisk. Apart from this, it drives the mobility of students and lecturers both inside and outside the campus.

Disaster Recovery

“Companies looking to get more “bang for the buck” from their VoIP implementation should use it to bolster their disaster recovery plans. VoIP needs to be at the core of any well-planned communications recovery strategy.” (Kerravala, 2007, para.1). VoIP helps in ensuring the disaster recovery of systems. At the time of disasters, VoIP can be helped for the establishment of continuous communications. Most of the disasters in the network can cause many troubles. The working of VoIP is at the third layer and it works like the other IP applications. For example, in the case of email, there is no requirement for the user to be concerned about the location of the mail server. The email functions when the user connects the plug to the Internet. The IP is required to function in this manner.

Security Issues

When the VoIP technology first came into the market, people were largely concerned about the issues like reliability, cost, functionality, etc. Now VoIP is used largely and there are also many security problems associated with it. Hacking is a major type of security issue that is faced by VoIP. Hacking enables any person to steal the services or use them when it is passed from one person to another person. There are chances of encryption in the VoIP calls which will cause the stealing of calls. Eavesdropping is a method by which the hackers can get names, passwords, etc which will help the hackers to gain access to private accounts. Viruses and worms are also a major security threat to VoIP networks. The VoIP phones, software, etc are susceptible to viruses, malware and other worms. VoIP is also susceptible to spamming. Sending mails to anyone without their permission is called spamming. Spamming is not very usual in VoIP now. But it is going to be common in the near future. The IP address in the VoIP accounts enables the sending of emails to several IP addresses. This makes the works of the spammers easier. Spam mails can result in the spreading of viruses. Some other issues like tampering of calls, denial of service, etc are also common in VoIP technology. Call tampering includes the corrupting of phone calls. The denial of service attack denies the services to the network. All these are major security issues that are expected to occur in the VoIP networks.


There is no doubt that VoIP has been a revolutionary technology that has bought significant change or reworks in the telephone industry. As what is stated in Moore’s Law, this technology had gained significant development, steady progress and gained immense popularity and has got the capability to replace the entire traditional phone system and that day is not so far.

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