the basics of voip

8/11/2019 The Basics of VoIP

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Running head: THE BASICS OF VOICE OVER INTERNET PROTOCOL (VOIP) 1

The Basics of Voice over Internet Protocol (VoIP)Spartak Goskolli

Hodges University

CIT 3215

Professor Chamas

November 30, 2013


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THE BASICS OF VOICE OVER INTERNET PROTOCOL (VOIP) 2

Table of Contents

Abstract ........................................................................................................................................... 4

Introduction ..................................................................................................................................... 5

Background Information about VoIP ............................................................................................. 5

What is Voice over Internet Protocol (VoIP)? ............................................................................ 5

How does VoIP Work? ............................................................................................................... 6

History of VoIP ............................................................................................................................... 6

Technologically-based Historical Information ........................................................................... 6

Culturally-based Historical Information ..................................................................................... 7

Economically-based Historical Information ............................................................................... 7

Common Types of VoIP Services .................................................................................................. 8

Analog Telephones ..................................................................................................................... 8

IP Telephones ............................................................................................................................. 9

Softphones .................................................................................................................................. 9

Video-over IP Applications ...................................................................................................... 10

Common VoIP Protocols .............................................................................................................. 11

Signaling Protocols ................................................................................................................... 11

Transport Protocols ................................................................................................................... 15

VoIP Quality of Service Assurance (QoS) ................................................................................... 16

Voice Traffic ............................................................................................................................. 16


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Loss ........................................................................................................................................... 17

Latency ...................................................................................................................................... 17

Jitter .......................................................................................................................................... 17

Services and Protocols .............................................................................................................. 18

Advantages and Disadvantages of VoIP ....................................................................................... 19

Advantages of VoIP .................................................................................................................. 19

Disadvantages of VoIP ............................................................................................................. 20

VoIP Security ................................................................................................................................ 21

Business Information, Implementation Cost and Equipment ....................................................... 23

Conclusion .................................................................................................................................... 23

References ..................................................................................................................................... 25


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Abstract

This document will depict information about the basics of Voice over Internet Protocol (VoIP).

The information will express what VoIP is, the history of VoIP such as when the first software

option was made available for purchase, information about how VoIP functions, VoIP Protocols

and the networking layers that they operates under, the advantages and disadvantages of utilizing

VoIP, the quality of service (QoS) involved in VoIP, the cost of implementing VoIP as well as

the equipment required to do so, the different types of security complications one may encounter

with VoIP, and why some businesses have chosen to utilize VoIP in their offices and corporate

buildings.


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The Basics of Voice over Internet Protocol (VoIP)

Introduction

Technology is progressing at a faster rate each passing day. Voice over Internet Protocol

(VoIP) is one of those technological utilities that is very heavily used in the business world. This

document will describe the basics of VoIP such as background information about what VoIP is

and how it functions. The document will also hold information in regards to the Technological,

Economical, and Cultural history of VoIP. Common types of services such as softphone and

video over IP, VoIP Signaling and Transport Protocols and the OSI model networking layers

they operate under, as well as Quality of Service Assurance responsible for the occurrence ofdata loss will be depicted. Although that seems to cover a large portion of the topographies of

VoIP, there is also the implementation side of this useful tool that individuals must consider.

This document will also contain information about the advantages and disadvantages of

implementing VoIP, as well as some common security complications one must be prepared to

encounter. To conclude the document, information about the cost of implementation, the

distinctive type of equipment required, and information about businesses and VoIP functioning

collectively will be specified. Although it may seem like VoIP is more a headache than a value, a

greater understanding will allow the individual to alter their opinion of this remarkable tool.

Background Information about VoIP

What is Voice over Internet Protocol (VoIP)?

According to Cisco Systems Voice over Internet Protocol can be defined as a

technological tool which allows a particular individual to make, as well as receive phone calls

over the internet (n.d.) VoIP uses the analog signal that a normal telephone utilizes and converts

it to a digital signal which can travel throughout the internet.


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How does VoIP Work?

In order to dispatch the voice data of human beings over the internet, the digital data must

be turned into packets. Packetizing a human voice involves attaching the routing information to

the packet header containing the voice data. Once an amalgamation of voice samples has been

pooled into a packet, it is dispatched through a multitude of switches and routers until it has

reached the designated endpoint, and is then decompressed. The decompression of the packets is

occasionally a seamless process that one does not detect unless there is a high volume of latency

within the network (Hallock, 2004).

History of VoIP

Technologically-based Historical Information

According to Hallock, in order for VoIP to have even been conceivable, the invention of

two very important technologies was mandatory: the telephone, and the internet. The first

telephone exchange occurred in 1878 in New Haven. In 1968 the internet was developed by

ARPANET (Advanced Research Projects Agency Network), which was an agency that the U.S.

Department of Defense established in 1957 (2004). Greenberg states that when the telephone was

devised the voice was sent as analog packets along a land line (2013). Once these technologies

had been implemented, a small establishment titled Vocaltec, Inc. released their invention known

as InternetPhone which permitted users to call one another via computer workstations providing

a microphone and speakers were present (Hallock, 2004). By 1998 VoIP calls had barley reached

1% of all voice calls made in the world and that number reached about 25% in the year 2003.

VoIP had a major advance in 1988. Providers began constructing software which translated

digital data into analog data. This allowed VoIP calls to connect with calls made on a public

switched telephone network (PSTN). The capability of connecting to PSTNs as well as the


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enhanced quality of broadband internet allowed VoIP to be operated to its full potential

(Greenberg, 2013).

Culturally-based Historical Information

Skype drew the attention of the populace during the release of its beta software in 2003.

Skype quickly became the standard for quality of video and voice communication over the

internet. The application allowed users to place phone calls as well as chat in videos over the

internet. Skype also allowed users to dial land line and mobile numbers (Thomas, 2010). Even

though Skype is mostly used by the general population, VoIP is also heavily implemented in the

business world. A large plethora of multinational organizations have begun using VoIP in orderto diminish the cost of communication exchange as well as streamline their infrastructure.

Hallock also declares that he does not pretend to be a technological determinist. And does not

think that VoIP unaided will modify human culture, but how the technology is used may (2004).

Economically-based Historical Information

The most prevalent amount of excitement in conjunction with VoIP is related to

economics. Although companies could depress the cost of their infrastructure spending, VoIP at

the time had much more shoddier quality than regular telephones. Problems with the Quality of

Service (QoS) surrounding VoIP led to the conception of Internet Telephone Service Providers

(ITSP). ITSPs were able to guarantee a certain quality by upgrading the current network

infrastructure, which led to the providers making a substantial amount of money. The biggest

advantage of VoIP is the capability of making international phone calls at a much lower cost than

conventional telephony services (Hallock, 2004).


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Common Types of VoIP Services

Analog Telephones

In VoIP, if an individual is using a traditional telephone, those telephone signals must be

converted to a digital format prior to being sent through a TCP/IP network even if the entire

VoIP connection is digital. The conversion of analog signals becoming digital signals involves

compressing and encoding the analog signals, which occurs at the presentation layer of the OSI

model. Although detailing the variety of voice codecs is beyond the scope of this document, the

necessary equipment to allow an analog to digital conversion to occur will be portrayed. One

way of assuring a successful conversion is by connecting an analog telephone adapter (ATA) to aVoIP adapter. In some cases the ATA is a card built into the workstation, or is an attached

device. The ATA allows one or more telephone connections, which are coupled through an RJ-

11 port on the adapter. The ATA then converts analog voice signals into IP packets and vice

versa. Another approach to achieving an equivalent outcome is by connecting an analog

telephone line into a switch or router which is capable of accepting analog signals, which will

then convert them into packets and relay them throughout the network. A third instance of

translating Analog signals into digital signals is a device known as a digital PBX, which is also

referred to as an IP-PBX (internet protocol private branch exchange). The PBX is a switch that

interprets both digital and analog signals, and converts them. This device can also be used to

connect with traditional PSTN lines and networks. There are also analog PBX devices that act in

the same manor, except the steps are executed in reverse order. Technicians can manage the PBX

settings through the specialized software which is bundled with the PBX. An additional

alternative is by utilizing a virtual PBX which is sustained on a service providers network. The


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benefit of using a virtual PBX instead of a physical PBX embraces the elimination of hardware

maintenance, upgrade, and replacement costs (Dean, 2013).

IP Telephones

Most VoIP infrastructures in this day and age use IP Telephones. IP Phones can only

transmit and receive digital signals. When using IP phones, the users voice is instantaneously

warehoused as digital data, which eliminates the necessity for analog conversion. Each IP phone

has its very own IP and MAC address on the network infrastructure, the IP and MAC address can

sometimes be managed by a Domain Name System (DNS). IP Phones still need to utilize a IP-

PBX in order to be transmitted throughout the network. IP Phones are very versatile and can berelocated from one network to another if an organization elects to change administrative center

locations. If an organization has opted to make the jump to utilizing VoIP, IP Telephony is a

prodigious approach to ensure a high quality of service, and can conserve monetary resources if

the organization decides to change the whereabouts of their networking infrastructure (Dean,

2013).

Softphones

Considering the different types of VoIP amenities may lead an individual to discover a

software solution that has telephone capabilities embedded within, without the physical device to

accompany it. This is known as a Softphone. Softphones provide the same calling functions as

traditional analog telephones or IP phones, however they are a software program that is installed

within your workstation, and uses the speakers and microphone connected to the system to

operate. Another option one can use when working with softphones is a headset. Softphones

connect to the network however they deliver services differently from IP phones. In order to

utilize a softphone, one must meet the minimum hardware requirements for the workstation such


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as a full-duplex sound card, a microphone, and speakers. If the system meets the minimum

requirements, one simply installs the IP Telephony client on the workstation. Instead of

physically dialing digits, the user is presented with a graphic user interface which comes with the

IP telephony client installed on the workstation, which allows them to input the desired phone

number. The benefits of using a Softphone include the extreme versatility which is ideal for

employees that do a sufficient amount of traveling. Softphones also have the ability to track call

information such as the date, time, duration, caller names, and the number. Softphones also

simplify recordkeeping and billing. An eminent case of a widely used Softphone is the Skype

software mentioned previously (Dean, 2013).Video-over IP Applications

According to Dean, Cisco Systems estimates that by the year 2015, over two-thirds of

internet traffic will be from video formats. The reason for this outstanding amount of progression

includes the large quantity of available video content online, as well as the accumulative number

of devices able to access the internet. Another motive fueling this growth is the large amount of

technological advances which are leading to the decreasing cost of equipment and bandwidth.

Technologies involved in Video over IP include IPTV, Video Streaming, and Video

Conferencing. The simplest application in the Video over IP classification is Video Streaming.

Video streaming has very rudimentary computer hardware and software requirements. The files

are stored on a server, which sends them out to the workstation that is requesting the stream.

Video streaming is very popular and can be viewed from an individuals web browser and can

also be a live stream. A big contributor to video streaming is the YouTube website. The next

application in the Video over IP category is IP Television (IPTV). IPTV is distributed by

telecommunication carriers, as well as cable company networks. If a user wishes to use IPTV


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they must have a High-bandwidth internet connection as well as a digital IPTV television signal.

The final breed of Video over IP technology is videoconferencing. Videoconferencing is used in

many fields of study including medicine, education, law, and information technology.

Videoconferencing is also used for surveillance software. If an individual wishes to utilize

videoconferencing they must find a method of generating, sending, and receiving audiovisual

signals. The workstation the user is on will require a camera, microphone, as well as the software

to administer and supervise the videoconference. Although these are the most common forms

Video over IP services, there are many other arrangements that an individual may choose to use

(Dean, 2013).Common VoIP Protocols

Signaling Protocols

The following signaling protocols only scratch the surface of VoIP; however, they are

very common. Both Voice over IP and Video over IP have a mutual transmission method.

Signaling is the term used to describe the information which is exchanged between each node in

the network infrastructure responsible for establishing, monitoring, or releasing those

connections, as well as controlling system operations. The set of protocols responsible for

managing the sessions between each client are known as the signaling protocols. Towards the

origination of the VoIP timeline, major corporations had their own proprietary signaling

protocols, meaning both users in the session were required to have the same hardware and

software or the communication connection would not be established. Today, the signaling

protocols used for VoIP have been standardized (Dean, 2013).

The first standard this essay will reference is the H.323 standard. The H.323 standard

refers to the group of protocols that are responsible for establishing multimedia sessions, as well


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as managing them. The protocols within the H.323 standard support both video and voice over

IP. The H.323 ITU standard can be broken down to five key elements; The H.323 terminal,

gateway, gatekeeper, Multipoint control unit (MCU), and zone. The H.323 terminal is described

as any node on the network that transmits visual, audio, and informational data to another node.

Any device that is responsible for issuing streaming video can be considered an H.323 terminal.

The device that is responsible for the translation between each device on the network that is

utilizing the H.323 signaling protocols as well as running other signaling protocols is known as

the H.323 gateway. The H.323 gatekeeper is responsible for authorizing and authenticating

gateways, terminals, as well as managing bandwidth and overseeing call routing, accounting, and billing. Although a very vital option to consider, the H.323 gatekeeper is a voluntary addition to

the H.323 networks. The computer system that is responsible for supporting and managing

multiple H.323 terminals as well the communication between them is known as the multipoint

control unit (MCU). An H.323 zone is a collection of terminals, gateways and MCUs whi ch are

managed by a single gatekeeper. The also H.323 standard contains the H.225 and H.245

signaling protocols. Both protocols operate at the Session layer or Layer 5, but each protocol is

utilized for a dissimilar function. The H.225 protocol is responsible for handling calls and

videoconference signals, and the H.245 protocol confirms that the type of data being transmitted

is formatted in a way which the H.323 terminal can interpret, whether that data is voice or video.

H.323 has been set up as an open protocol by ITU for multiservice signaling (Dean, 2013).

Another protocol that was released after the creation of H.323 that enticed a surplus of

attention from networking specialists is known as the Session Initiation Protocol (SIP). The SIP

performs functions very similar to those within the H.323 standard; however SIP is a control

protocol for multiservice networks which performs under the Application layer, also identified as


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Layer 7. SIP has been developed to model the Hypertext Transfer Protocol (HTTP). An example

of this can be seen when text based messages that are exchanged by clients to initiate VoIP calls.

The text based messages are formatted similar to a HTTP request and relay on URL addresses.

The developers of SIP sought out to reutilize as many of the prevalent TCP/IP protocols as they

possibly could for managing sessions as well as providing enhanced services. SIP is capable of

determining the endpoint location, which is referred to as any gateway, server, or client that is

communicating on the network infrastructure. SIP also determines the availability of an endpoint

device. If a client is unreachable, SIP will return a message which states whether the client is not

responding or is already connected to a call. SIP is also accountable for session establishment between two endpoint devices and managing transferring, adding, and the dropping of call

participants. SIP manages the features of a call or videoconference. SIP will either agree on a

type of encoding for all endpoint devices on the call, as well as manage the feature changes while

the endpoints are connected. The H.323 groups functions are far less limited in comparison to

SIPs functions. Similar to the H.323 group, SIP has a collection of terms used to specify the

architecture of the network. The components included in an SIP network are the User Agent,

User Agent Client, User Agent Server, Registrar Server, Proxy Server, and the Redirect Server.

The User Agent refers to any node on the network infrastructure that is capable of responding to

SIP requests. The User Agent Clients are end user devices which instigate a SIP connection.

These devices include and are not limited to workstations, smartphones, IP Telephones,

computers, and tablets. The User Agent Server is a type of server that responds to the session

requests of User Agent Clients. The User Agent Server is responsible for initiation of the session

as well as termination. Both User Agent Servers and Clients are considered User Agents. The

database containing the location information of each user agent is stored on the Registrar Server.


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The Registrar Server also audits which domain each User Agent resides within. The Proxy

Server determines the location of the User Agent that is requesting a connection and elects to

either pass the information to a Redirect Server, or to query the nearest Registrar Server on

behalf of the User Agents. If the User Agents requesting a connection reside within the SIP

Proxy S ervers domain, the server acts as a go -between for all established and terminated calls. If

the User Agent is not within the Proxy Servers domain, its information is passed to a Redirect

Server on the network. Proxy Servers are an optional addition to the SIP network infrastructure.

As a final point, the SIP Redirect Servers are responsible for accepting and responding to

connection location information requests from Proxy Servers as well as User Agents. TheRedirect Server is not involved in maintaining or establishing sessions. Redirect Servers are also

an optional part of an SIP infrastructure, similar to the Proxy Server. Many VoIP vendors prefer

SIP over H.323 because it is very unproblematic to maintain, and requires far less instructions to

control a call. SIP also utilizes much less processing resources and in some cases is more flexible

than H.323. SIP is designed to endure work with many different types of Transport Layer (Layer

4) protocols. Popular SIP equipment providers include 3Com, Nortel, Avaya, Cisco, and Asterisk

(Dean, 2013).

Although H.323 and SIP are responsible for regulating call control and signaling for

VoIP or video-over IP clients and Servers, they do not get involved with the communication

between the media gateways. This communication is managed by two types of protocols: MGCP

and MEGACO. Media Gateways have the responsibility of converting the analog signal from

PSTN lines into VoIP format. Because media gateways are not responsible for regulating

signaling and call controls, they can make encoding, decoding, as well as translating data the

primacy of their resources, which in turn results in a faster processing of information. An MGC


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is referred to as a media gateway controller. This is a system which manages a multitude of

media gateways. MGC is also responsible for managing and disseminating information about

paths taken by video and voice between gateways. MGC performs the call switching functions

because it is software, which is why MGCs are at times referred to as softswitches. When media

gateways receive calls, they simply contact the MGC by sending a message alerting it about the

calls. The MGC then decides which media gateway will be used to translate the signal. The

MGC also determines the location of the physical media the call should be sent to. Once the

MGC has processed the information, it instructs the media gateways about what to do with the

call. Because of this, MGCs are extremely advantageous on larger VoIP network infrastructures.The MGC uses the Media Gateway Control Protocol (MGCP) to communicate with the media

gateways on the network through several types of protocols. MGCP is an archaic protocol that

can operate along with H.323 or SIP signaling and control protocols. MEGACO is a more recent

set of gateway control protocols that perform equivalent tasks as MGCP but uses different

commands and processes to accomplish them. Many networking specialists prefer MEGACO

over MGCP because it can support a superior assortment of networking technologies such as

ATM (Dean, 2013).

Transport Protocols

Although the protocols above are utilized to communicate information, a different set of

protocols are exercised at the Transport Layer (Layer 4) to deliver the video or voice

consignment. TCP and UDP also predominantly operate in the Transport Layer. TCP is a

connection oriented protocol and provides a measure of assurance when delivering information,

UDP is connectionless and does not. In VoIP and other real-time applications, UDP is preferred

because it can transport packets more rapidly due to the lesser amount of resource expense.


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There are two very common transport protocols that will be portrayed within this document: RTP

(Real-time Transport Protocol) and RTCP (Real-time Transport Control Protocol).

RTP is a protocol which resides in the Application Layer (Layer 7) which relies on UDP

that is housed at the transport layer to send VoIP packets to their designated terminus. RTP

applies sequencing numbers and a time stamp of when the data was sampled to packets which

indicates the order they should be assembled in when they reach their destination. Although RTP

applies the sequencing numbers, it does not identify whether the transportation was successful.

RTCP is the protocol which warrants whether or not packets have arrived at their destination.

RTCP is an optional tool within networking infrastructures that works with RTP. RTP and RTCPmay be able to provide information about packet loss, order and delay, but they cannot correct

transmission flaws when they occur. Attempts to correct transmission flaws are handled by

Quality of Service Assurance protocols (Dean, 2013).

VoIP Quality of Service Assurance (QoS)

Voice Traffic

Voice traffic refers to the amount of calls made on a VoIP network infrastructure. The

key QoS requirements and recommendations to ensure respectable quality VoIP service include

DSCP EF voice traffic marking, a Loss percentage which is less than one percent, One way

latency should not exceed 150 milliseconds, the average one-way jitter should be targeted at less

than 30 milliseconds, and a guaranteed priority bandwidth range of 21 to 320 kilobits per second

is required per call. Loss, Latency, and Jitter are factors that directly affect the quality of voice

calls (Szigeti & Hattingh, 2004).


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Loss

During calls - skips and voice clippings are both caused by Loss. Packet Loss

Concealment (PLC) is used to mask the effects of VoIP packets that encounter Loss. Every PLC

method is different and the one used depends on the type of codec. Waveform codecs use a

simple method that replays the last sample with increasing attenuation at every repeat. This

technique can be effective at some instances of Loss. The loss of two or more packets yields a

noticeable difference in voice quality, which is why it is recommended that a loss rate of less

than one percent is present (Szigeti & Hattingh, 2004).

LatencyIn VoIP, Latency is described as the amount of time it takes for the sound to travel from

the individual who articulates it, to the individual who is receiving it. Three types of latency exist

in VoIP: Propagation delay, Handling delay, and Serialization delay. Propagation delay is caused

by the distance a signal must travel as light through a fiber optic cable or electricity in copper

based wire; Handling delay refers to the amount of time it takes for compression and

decompression of packets (packetization) as well as packet switching to occur and is also known

as processing delay; Serialization delay is the amount of time it takes to place data (a bit or byte)

onto an interface. Serialization delay influences overall latency at a fairly minimal level

(Davidson, Peters, Bhatia, Kalidindi, & Mukherjee, 2006).

Jitter

According to Davidson, Peters, Bhatia, Kalidindi, & Mukherjee; Jitter is the variation of

packet interval time. Jitter is an issue which is only prevalent in packet-based networks. The

difference of when the packet is expected to arrive versus when it actually is received is called

Jitter (2006). A good way to prevent Jitter is by utilizing a Jitter Buffer. A Jitter Buffer is also


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but less popular in larger infrastructures. Bigger networks prefer more streamlined Quality of

Service techniques (Dean, 2013).

A great example of a streamlined technique used on larger networks is DiffServ. DiffServ

is used to prioritize traffic contained within the network. It modifies the IP datagrams that

contain payload data. When prioritizing traffic, DiffServ inserts its information within the

DiffServ field in an IPv4 Datagram. In IPv6 Datagrams Differentiated Service information is

placed within the Traffic Class field. DiffServ utilizes two diverse forwarding types: Expedited

Forwarding (EF) or Assured Forwarding (AF). When DiffServ is utilizing EF, the stream is

designated a minimum departure rate. When AF is utilized, one can assign different levels ofrouter resources to the data stream. Although AF prioritizes data, it does not guarantee that

packets will arrive on time or in sequence. DiffServs simplicity as well as its low overhead

makes it much better suited for large networking infrastructures (Dean, 2013).

The final QoS technique which belongs in the Network Layer (Layer 3) is known as the

Multiprotocol Label Switching (MPLS) technique. MPLS is used to indicate where data is to be

forwarded. MPLS does this by replacing the IP datagram header with a label at the first router a

data stream encounters. The label contains information about the location the packet is to be

forwarded to. MPLS forwarding is very agile because a router knows exactly where to send the

packet (Dean, 2013).

Advantages and Disadvantages of VoIP

Advantages of VoIP

Although all of the technical information about VoIP is enlightening to communicate,

one of the largest factors of information an individual or organization may be thirsty to collect

are the advantages of implementing VoIP. According to ProgrammerWorld, the leading


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contributors to the VoIP advantages are the low taxes, and the flexibility. Due to the calls being

made over the internet, governments all over the world are not heavily taxing VoIP services in

comparison to the local telephone companies. Also, when choosing a VoIP service provider, you

will be provided with a converter which is used to allow regular telephones to be used with VoIP

services. With that being said, an individual can take their phone converter, as well as their

phone number, and use them virtually anywhere in the world which has access to a high-speed

internet connection (2012). Another big advantage to VoIP is the low service cost. The cost per

minute and monthly fee of a VoIP phone service is much lower than that of the traditional POTS

(Plain Old Telephone Service) provided a local PSTN. VoIP service providers also allow theusers of their service access to a much larger pool of features than the POTS. Voice mail, call

waiting, call forwarding, and caller ID are a few features that are generally included with VoIP

services free of charge (Kochmanski, 2012). Although the cost of VoIP services is much less

than PSTN prices, Posey states that the cost of the service heavily depends on which provider

one chooses. Another one of the biggest advantages to VoIP is simplicity. Utilizing the calling

feature on a VoIP telephone is no different than a standard analog telephone (2006). Although

there are many advantages to VoIP, without being able to properly manage the disadvantages,

the services become virtually useless.

Disadvantages of VoIP

Although there are many advantages to using VoIP, there are also many disadvantages.

One of the biggest disadvantages to VoIP is security. This is a problem that is prevalent in most

internet technologies (ProgrammerWorld, 2012). The next section of this document will give

detail to the security issues of VoIP. Another disadvantage is that in order to use VoIP you must

purchase a VoIP adapter or ATA in order to use the SIP trunks with an already existing business


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phone system. An organization also runs the risk of having potentially inferior customer service

when choosing to use VoIP. When faced with an issue, the organization generally needs to hire

their own troubleshooter in order to repair the problems. Also, the utilization of fax machines

may be more difficult. One will have to occasionally deal with incomplete outgoing and inbound

calls, as well as dropped calls. The QoS of VoIP is not nearly as good as that provided through

the POTS line. There is also a lack of redundancy when implementing VoIP which is not an

issue with traditional telephones (Kochmanski, 2012). The biggest issue with VoIP is the voice

quality. A lot of the problems with voice quality have to deal with how the process of VoIP

works. When the files are broken up into sequential packets, they are sent in an unsystematicorder and there is a slight sum of latency when receiving the packets because they must all be

present prior to the packets being decompressed. Compression and decompression also utilizes

more processing power, which hinders the sound quality of the voice data being transmitted.

VoIP telephones also draw power from an external source, making them useless during a power

outage. Dependability is a big issue with VoIP services because if either power or an internet

connection is not prevalent, the phone services will not function. Finally, if an individual were to

dial 911 with a VoIP system, they may not be connected to a local dispatcher, and the location of

the originating call signal may not be traced depending on the location of the individual as well

as the IP address of the phone system (Posey, 2006). Although there are many disadvantages to

VoIP, if an organization is properly able to manage the issues, the service can be very beneficial

monetarily.

VoIP Security

The most important aspect that cooperate executives should certainly not overlook when

deciding whether to implement VoIP is the security threats. At the top of the VoIP security


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threats list are DoS (Denial of Service) attacks. DoS attacks target SIP extensions which exhaust

resources and lead to busy signals or disconnects. Another spam related threat is known as Spam

over Internet Telephony (SPIT). SPIT systems are similar to botnets. They target millions of

users by sending junk mail, which in turn clogs voicemail boxes and decelerates the systems

performance leading to a loss in efficiency. Directory Harvesting attacks are also a major threat

in VoIP because they can lead to hackers amassing a compromised list of the VoIP subscribers

by using brute force methods until the server validates a VoIP address. These subscribers then

become subject to future attacks similar to SPIT. The corporate executives must also recognize

that similar to email Phishing, there is Voice Phishing. Voice Phishing is referred to as Vishing;which attempts to divulge sensitive, as well as personal information such as account numbers,

user names, and passwords. VoIP networks are also subject to man in the middle attacks.

Hackers can spoof the MAC address of two users, and force VoIP packets to be redirected to the

hackers systems; this creates a problem because the hackers can reassemble those packets and

listen into the conversations, which can lead to other compromised VoIP System information

(Needle, 2009). Although there are many security problems to deal with, organization leaders

can rest assured. There are many solutions to assist in combating these issues. One can install

intrusion detection systems, VOMIT (Voice over Misconfigured Internet Telephones) tools, and

also add address translation methods to their networking infrastructure (Ransome &

Rittinghouse, 2005). With all of those security risks posing a big threat to companies, they are

still uncertain as to whether or not implementing VoIP as part of their network infrastructure,

even though there are a vast amount of tools that can combat the security threats.


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Business Information, Implementation Cost and Equipment

Although VoIP is very beneficial, many business executives need to evaluate the cost of

implementing VoIP. According to Gareiss, when implementing VoIP one must look at the cost

of the complete project. Companies spend about 20% more on the first two years of VoIP

implementation than they normally would on a TDM (Time-Division Multiplexing) system. The

executives must also evaluate the cost of switches, gateways, hand-sets, applications, and other

end devices. The cooperation may also be required to upgrade their Local Area Network (LAN)

because VoIP requires power over switches. The expenses are enormous, and that does not even

take into account the training of employees, equipment licensing and maintenance, monitoringand management tools, and future ongoi ng costs (2009). In Cisco Systems article How Voice

over IP (VoIP) works it is stated that implementers of VoIP will actually save on cost due to

having to manage one network instead of two because one can easily change phone extensions,

add, move, and change locations, which in turn will save money and give the corporation more

flexibility (n.d.). Although there are many different opinions about whether VoIP is a good

investment, one can always research and make the decision on their own based on their business

needs and growth projections.

Conclusion

In this document, basic information about VoIP was described. Although it may seem

like this is extensive, VoIP is a vast technology that has much more information to be researched

before one can truly consider themselves an expert on the subject. Information about what VoIP

is, which was defined as a tool to receive phone calls over the internet, as well as how it works

by describing the process of packetization, compression, and decompression. The technological,

cultural and economic histories of VoIP were also covered and the conclusion was drawn that


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without the invention of the telephone and the internet, VoIP would not exist. Also, todays

culture heavily relies on Skype and YouTube for video streaming as well as keeping in contact

with each other via video chat. This paper contained information about the common types of

VoIP services including: Analog telephones using an ATA to convert the signals to a VoIP

format; Softphones which use the internet as well as a headset and a software suite to

communicate using VoIP, and Video-over IP applications such as IPTV, Videoconferencing, and

Skype were discussed. Once the foundation of hardware basics was laid down, this document

provided information about the common VoIP signaling and transport protocols. The signaling

protocols that were discussed were H.323, SIP, MCGP and MEGACO; and the transport protocols that where discussed include RTP and RTCP. Research stated that even though these

protocols are used to make sure data arrives at the designated location, they cannot fix any

problems that occur. The Quality of Service Assurance section depicted common variables and

protocols used to determine the quality of the VoIP transmissions. After the foundation of

technological knowledge was established, the advantages and disadvantages of VoIP were

presented. An example of an advantage for VoIP is the large savings in cost, however a

disadvantage was the possible security threats one may run into such as DoS, SPIT, and

Directory Harvesting attacks. Business information, implementation cost, and equipment as well

as network upgrades and maintenance cost was discussed as well. Although there is a vast

amount of information to grasp, VoIP can be very useful for large corporations, but seems more

trouble than anything in smaller organizations.


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References

Cisco Systems. (n.d.). How voice over ip (voip) works . Retrieved from http://www.cisco.com/

Cisco Systems. (n.d.). What is voice over ip? the basics . Retrieved from http://www.cisco.com/

Davidson, J., Peters, J., Bhatia, M., Kalidindi, S., & Mukherjee, S. (2006). Voice over ip

fundamentals . (2nd ed.). Indianapolis: Cisco Press. Retrieved from

http://www.ciscopress.com/

Dean, T. (2013). Network guide to networks . (6th ed., pp. 555-580). Boston: Course Technology,

Cengage Learning.

Gareiss, R. (2009). The true cost of voice over ip . Retrieved from http://www.avaya.com/Greenberg, R. (2013, June 3). The history of voip . Retrieved from http://www.thedigest.com/

Hallock , J. (2004, November 26). A brief history of voip: Document one the past . Retrieved

from http://www.joehallock.com/

Kochmanski, D. (2012, July 16). Voip advantages and disadvantages . Retrieved from

https://www.google.com/url?sa=t&rct=j&q=&esrc=s&source=web&cd=2&cad=rja&ved

=0CEUQFjAB&url=http://www.ddktele.com/info/VoIP_Advantages_and_Disadvantages

_by_DDK.doc&ei=nVSaUvHnJ8ndkQeor4H4Ag&usg=AFQjCNGsAEIdpNJaJpseJ3Cw

aPhjxNJ-3w&sig2=8sudLoY7ZjfGtSpe0o8YMw&bvm=bv.57155469,d.eW0

Needle, D. (2009, April 16). Top voip threats detailed by security firm . Retrieved from

http://www.internetnews.com/

Posey, B. (2006, August 2). The pros and cons of voip . Retrieved from

http://www.windowsnetworking.com/

ProgrammerWorld. (2012). What are the advantages and disadvantages of voip . Retrieved from

http://faq.programmerworld.net/


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the basics of voip

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