Voice over IP (VoIP) also known as Internet Protocol (IP) telephony refers to a form of communication that makes it possible for an individual to make calls over internet connection broadband instead of using analog telephone lines. Normally, VoIP access allows an individual to call another who receives the call over the internet. These services are referred to as interconnected VoIP and they allow someone to call others such that they can receive calls to and from traditional landline numbers, often, for service payment. Some of the VoIP services require a devoted VoIP computer or computer while others permit an individual to use their landline telephone while placing the calls through a specific adapter. As such, VoIP term which is derived from IP telephone stands for a set of facilities used for purposes of controlling voice information delivery over the internet. The major significance of VoIP is that it avoids high fees charged by the normal providers of telephone services. The technology is rapidly growing for audio communication and it uses ubiquity of IP based networks while setting up VoIP consumer devices such as VoIP getaways, desktop IP and mobile VoIP enabled portable devices in businesses and majority of homes throughout the globe. Devices that are VoIP enabled include portable IP phones, VoIP gateways and desktop. Such devices decrease fee data and voice communication, improve the existing features as well as incorporate the latest compelling communication features and data services (Chiang, 2006).
VoIP comes from VoIP Forum, an endeavor by foremost equipment providers like 3com, VocalTec and Netspeak for purposes of boosting use of ITU-T H.323. This refers to the standard that is used to send voice and video via use of an IP within an intranet and public internet. The forum, in addition boosts use of standards of directory services in order to ensure users can trace other users and use touch tone signals for purposes of automatic voice mail and call distribution. VoIP, as well uses RTP (real-time protocol) to make it ensure packets are received in a fashion that is timely. Using public networks, VoIP is currently difficult to guarantee Q0S (Quality of Service). However, it is possible to enjoy better services with private networks which are managed by Internet Telephony Services Provider (ITSP) or by enterprises. VoIP has become a major primary driver towards audio communications evolution. The technology, additionally is significant not just for mobile phones but the broad application platforms which facilitate audio interactions on devices like desktop computers, mobile devices, set-top boxes as well as other devices with software that is specific to certain companies where audio communication is an extremely vital feature (Ghencea, 2007).
While presently VoIP offers a small portion of the income, it is developing rapidly. Research estimates wholesale VoIP sales will advance to more than $ 400 million in 2002. The overall equipment sales of soft switches such as IP Private Branch Exchange (IP PBX), VoIP gateways and VoIP application software were as well anticipated to reach close to $ 12 billion by the year 2006, a 6 fold growth figure over 2001. In the same manner, income from selling wired enterprise IP phones was anticipated to go beyond $ 2.7 billion by the year 2006 in which case the figure did not comprise of mobile IP phones or phones used in personal homes. Unlike in the case of long-established cellular networks and public switched telephone networks (PSTNs), the Interne was not initially intended for real-time dedicated network for audio communications. The internet, however was created as a data communication asynchronous, which made it possible for data packet retransmission and loss without dedicated bandwidth for each user. Besides, VoIP is not controlled by any one, centralized operator with the ability to manage flow and quality of caller communications. Unlike in the case of cellular networks and PSTNs, the VoIP is comprised of service providers and disparate networks which add to the complexities of provision of real tie integration services. This factors integration makes VoIP a tricky medium network for situations of real time communication like audio communications. However, despite these difficulties, VoIP type of internet is a keystone of the Internet since it continues to move on real time centric networks. Also, the internet provides 4 primary benefits which include:
- Compelling services enhancing consumer productivity
- Reduction in total ownership fee
- Well organized use of network
- Great operational flexibility
Types of VoIP Technologies
Simply, soft phone refers to a program that runs on computers. The program makes it possible for one to communicate via a microphone that is connected to a computer and they listen via headphones or computer speakers. There are different kinds of softphones available and majority are issued as freeware. Generally, freeware types make it possible for one to talk with another person so long as they have a similar program on their computer. Instead of connecting through a normal phone number, one chooses who they want to speak to from the drop-down menu which is on the computer screen or they enter the IP number of the computer. In this case though, a broadband connection is needed even though one chooses to use freeware soft phone only. Even though there is the chance of talking using a dial-up connection, the audio connection is often of low quality with irregular tone and significant distortion. Some of the soft phones as well provide the ability of calling other users on normal phones though this is often a free-based system where an individual pays per minutes spent. However, while using soft phone with a computer, one is able to get a call that is of better quality through use of unique VoIP headset instead of relying on computer and microphone speakers. The soft phone that is computer based is simpler to use and install since it does not require any extra software. The interface as well is quite simple and often appears just like a phone keypad on computer screen. In an environment that is business oriented, one can install VoIP system for the office. Majority of the VoIP carriers target small business providing multiple extensions; though large offices can as well set up an IP PBX. Other carriers as well provide hosted IP PBX solutions with excellent features that some of the largest organizations that install conventional PBX’s are unable to match. The IP PBX’s solution aspect is the ability to connect several phones, combine branch offices or telecommuters into a single virtual office where each, is comprised of an extension of key phone number, despite their geographical position (Ismail, 2012).
These refer to devices that permit someone to use traditional phone to replace VoIP calls. These devices are connected to DSL/cable modems and they permit one to attach to normal telephone. When the devices are configured correctly with an ideal VoIP service plan and provider, they do not need any special interaction or program with a computer. What users need is to pick their phone and dial the number at dial tone. One can also carry the adapter in order to facilitate making calls anywhere there is accessible broadband (Mehta, 2001).
These are devices facilitating use of traditional phone while placing VoIP calls. Usually, they are in the form of ESB adapters which are larger as compared to the normal thumb drive. They also have standard modular phone jack features where the ordinary phone lines are attached. When they are connected, the phone serves as if it were connected to a standard phone service.
Table 1. Comparison of VoIP devices
|Desktop||Most of these devices are found in an office desktop which have either graphical or text display, for example two line LCD screen, QVGA9 touch screen composed of buttons and numbers.
Generally, IP screen phones have broader features set as compared to those that do not have. For instance, screen phones might include browsing and capabilities of sending messages instantly. Added hardware requirements make screen phones to me more expensive.
|Mobile||Mobile devices are normally suited to businesses where there are wireless technologies like 802.11 networks. The devices are grouped into two subgroups, which are;
|Base station||Some types of VoIP devices are stationary while others combine both mobile VoIP device and stationary base station. This base station or rather cradle might provide extra functionality like local storage for the voice mail as well as highly developed calling features. Additionally, the base station may be part of home server, gateway or set-top box. It can be part of game console which connects IP end points to an IP networks or connect regular PSTN (public switched telephone network) phones to IP networks.|
|Conference room||VoIP devices for instance, IP conferencing phones, may be used for calling two or more participant groups or participants. In public places, conferencing phones might include security features like login authentication, which allow access to only particular users groups. Similarly, diverse calling privileges may be associated with particular groups. For instance, group A can be allowed to call international numbers but for group B might not.|
Chart 1. Different types of VoIP services and how they work
Megaco (H. 248)
Media Gateway Control Protocol (MEgaco) is the result of combined efforts of ITU-T and IETF study group 16. The protocol is one that is used between elements of a physically disintegrated multi-media gateway. The network packet interface might include AT, IP and others. Such interfaces support varying types of SCN signaling system which includes tone signaling, QSIG, GSM, ISDN and ISUP as well (Mohamed, 2016).
Media Gateway Control Protocol (MCGP is applied in controlling telephony gateways from external call control constituents which are known as media gateway or call agent controllers. The gateway telephony technology is a network constituent offering conversion between data packets which are either passed over the internet or other varying voice signals and packet networks passed on telephone circuits and data packets over the internet or over other packet networks. The protocol is one that presumes call control structure in which case the call management intelligence is out of gateway and controlled by outer call control components. The protocol makes the assumption that call control component or call agent is in actual sense, a slave protocol where the gateway is expected to run on commands which are propagated by a call agent.
These are set of standards which collectively are referred to as Multipurpose Internet Mail Extension. They are as well defined as messages format that permit textual bodies in the form of character sets rather than of Us-ASCII, which is a wide set of formats that are dissimilar for the multi-part message bodies, non-textual message bodies and textual header information in character sets besides US-ASCII. The first standard which is in the set RFC 2045 identified different headers which are used to structure MIME messages. The RFC 2046 explains common structure of MIME media typing system as well as defines the first media types set. RFC 2047 is the 3rd standard which describes extensions to RFC 822 that facilitates non-US ASCII text data in the internet mail header fields. RFC 2048, the 4th standard is responsible for specification of different IANA registration process for facilities that are MIME related. RFC 2049 is the 5th and last standard that describes MIME conformance criterion and provides illustrative examples of Mime message bibliography, formats and acknowledgments. RFC 2045, which is the initial standard also defines the header fields’ number which includes the type of content. Content type refers to a field that is used to identify nature of data in the body of MIME entity by assigning media type and subtype identifiers and it also provides auxiliary information which is needed for certain media types (Mouchtaris, 2006).
The Remote Voice Protocol refers to MCK communication protocol which transports digital telephony sessions over packet or circuit based data networks. The protocol is mainly used in the MCK’s Extender product family broadening PBX services over WANs (Wide Area Networks). The protocol also facilitates the need to establish connection and configuration between the server device and client device. RVP uses TCP to transport control data, signaling as well as UDP for transport of voice data.
A decoder or coden handles the conversion of signal from digital forms and vice versa. VoIP technology as well uses a variety of decoder for voice, video or the two. In VoIP the decoder used is normally known as payload type for RTP packet or encoding method. The designers of the codec look forth to optimizing on three major factors. These include; decoding/encoding speed operations, the fidelity and quality of the video/sound and the size of consequential encoded data stream.
Table 2. VoIP comparison
|Codec||Data Rate||Packetization Delay||Bandwidth|
|G.711u||64.0 Kbps||1.0 msec||87.2 Kbps|
|G.711a||64.0 Kbps||1.0 msec||187.2 Kbps|
|G.726||32.0 Kbps||1.0 msec||55.2 Kbps|
|G.729||8.0 Kbps||25.0 msec||31.2 Kbps|
|G.723.1 MPMLQ||6.3 Kbps||67.5 msec||21.9 Kbps|
|G.723.1 ACLEP||5.3 Kbps||67.5 msec||20.8 Kbps|
Significance of VoIP
Compelling Latest Services that Facilitate Consume Productivity
The extensibility and flexibility of VoIP enables service and software providers as well as manufactures to offer more successful expose to a wide variety of services and features that add direct value to consumers. The combination of video, voice and data with IP networks and devices makes the interaction and exchange of these kinds of interaction easier, as such enhancing consumer productivity. Some of the commonly value added features and services for consumers include:
- Interactive audio recognition
- Unified messaging
- Call center administration
- Voice mail
- Conferencing services
- Voice mail
- User relationship
- Database queries such as e-mail lookup
- Instant messaging and web browsing
Innovations yield functionality that is improved and value added services. For example device manufacturers and service providers currently using VoIP software as competitive differentiators while maintaining and obtaining users. A major milestone in Microsoft approach for creation of VoIP rooted systems are Microsoft Windows CE, which refers to a modular and flexible application platform that is easily configured and extended to offer a functionality that is marketable such as e-mail, instant messaging, web browsing, video capabilities and e-commerce. Some of these services that are VoIP related and features make users’ live in a manner that is more efficient while others make it possible for companies to be more convenient. For example, research shows unified messaging makes it possible for consumers to use single systems to access any form of message such as voice mail, video or fax and e-mail saving a network up to twenty five minutes each day.
Decreased Total Fee Ownership
VoIP technology facilitates he combination of audio communication traffic as well as data traffic into individual network as such, reducing overall fee of ownership which is associated with the integrated data/voice network. By using VoIP, analog audio signals, are digitalized and then converted to data packets which get propagated over IP based networks. The combination of different media types like voice, data and video into a single network eradicates the infrastructure and maintenance redundancies as such, helping to reduce capital and operation costs. Another importance of using the network on its own for voice, data and video dissemination is that there are numerous elements such as client devices, call servers and application servers for audio mail storage that can be integrated easily. On the other hand, server/client services that are advanced permit VoIP systems and devices to be provisioned and managed remotely. Distant management decreases costs which include expenditures associated with consumer Moves, Adds and Changes (MACs) and costs related to upgrading of edge devices with the latest user applications and services. VoIP systems also progressively demonstrate large cost effectiveness as compared to traditional audio networks. Because VoIP develops cost/benefit ratio together with efficiency and flexibility in implementation, it will continue to develop.
Besides low network fees, VoIP system users and enterprises also benefit from savings in long distance charges, service fees and bundle services. Examples are as highlighted:
- Unlike available proprietary networks of telephone service long established sources, VoIP is an open network that can be used in different businesses that offer voice services. As such, being an open network, VoIP encourages competition among service providers leading to low service costs for individuals and organizations.
- Communication networks such as cable networks and satellite can also be made to provide VoIP services. As such, cable and satellite providers package VoIP services with the content access services and data subscription available such as cable TV subscriptions as such, offering users a range of service bundles accompanies by new discounts.
Usually the Internet fails to distinguish state or country borders; however, with VoIP systems, the distinction among international, long distance and local calling often disappears and the consumers are able to save on long distance and international fees (Thomsen, 2000).
Efficiency Network Utilization
VoIP technology makes provision of significant gains towards efficiencies of bandwidth that in turn, decrease fees as well as increase service quality. Various factors also contribute to effective use of bandwidth for VoIP systems. These include:
Silence elimination: PSTN is based on (TDM) time division multiplex technology. In the network, interaction capacity is allocated to a consumer even at times when the user is not talking. Close to 50 percent of standard voice communication is silence. As such, this means almost half of the capacity of TDM network remain unused due to silence only. In VoIP technology, capacity is not allocated constantly but rather is made accessible as defined by technology.
Redundancy Reduction: Close to 20% of human communication is comprised of recurring patterns. The predictable TDM network does not use the techniques of reduction in elimination of recurring audio signals. Such lessening techniques are universal in VoIP technology.
Data Throughput Efficiency: The complex analog to digital operations that VoIP technology uses provide a throughput that is far efficient compared to similar operations used in the traditional TDM network. As such, redundancy reduction, silence elimination and more proficient data throughout and others, VoIP uses close to 10 to 15 percent of bandwidth that is needed for traditional audio communication.
Better Operational Flexibility
Another reason that explains the increasing VoIP technology is adoption of fundamental technology which is far flexible and extensive compared to traditional audio broadcast technologies. In the circuit-switched established transport, voice networks, application latters and call control are classes into one proprietary systems. However, in VoIP systems, such divisions are desecrated to varying components where everyone can be integrated or substituted as required in the entire system. Such disintegration makes it possible for the services and system to be dynamically controlled and designed. The movement from proprietary, end to end solutions to combine, vendor centric, open, IP based environments leads to much flexible, customized and extensible systems. Technologies that are VoIP related such as extensible (SIP) Session Initiation Protocols are based on the fact user preferences like client devices, underlying services and infrastructures vary from time to time. For example, video conversations and voice calls arise between infrastructures which apply different hardware components. On top of this, SIP is scalable and flexible as such, ideal for use in different applications, scenarios and infrastructures. SIP also offers platforms that are identical such as video, instant messaging voice and presence information. The long established division between network to user and network to network on the other hand have become outdated with SIP, leading to simple interoperability between reduced operation fees and separate systems.
Chiang, W. (2006). A performance study of VoIP applications. Journal of management information technology. 230-400.
Ghencea, A. (2007). QoS and Voice Over IP. Journal of Knowledge Management, Economics and Information Technolog.23, 56-150.
Ismail, M. (2012). Performance analysis between IPv6 and IPv4: Voice over IP implementation in campus network. 109-200.
Mehta, P., Udani, S. (2001) Voice over IP. IEEE Potentials (Vol. 20, pp. 45-98).
Mohamed, S. (2006). Performance comparison of packet transmission over IPv6 network on different platforms. IEE 153(3):425-600.
Mouchtaris, P. (2000). Voice over IP signaling: H.323 and beyond. IEEE communications magazine (Vol 38, pp. 46-120).
Thomsen, G. (2000). Internet telephony: going like crazy (Vol. 37, pp. 67-140).