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Discussion

Write an essay on Voice Over Internet Protocol?

Voice over Internet Protocol (VoIP) is a name used in Internet Protocol (IP) in order to communicate as an alternative of conventional analog system (Dickinson, Marshall  & Helme, 2015). A usual connection of phone is needed for VoIP facility where as others allowed you only by using the internet connection you can make your calls through telephone. A benefit is that the normal telephone charges are avoided in VoIP and internet telephony. Fundamental VoIP admission generally permits you to call others who are also getting calls in excess of the internet

“Internet phone” is the primary VoIP function, introduce in 1995, developed by an Israel company named "Vocal Tech”. This application was basically run on Personal computer (PC). The tone signals were first compressed and then converted into IP packets for communication through internet. Delay, disconnection, low quality and inappropriateness are some of the disadvantages of “first generation” VoIP.

VoIP is a cover picture for any examination for delivering usual voice phone examine through internet protocol (IP) (Jackson et.al 2014).

“Voice over IP is the tools of digitizing noise, compact it, flouting it up into data package, and deliver it in excess of an IP network where it is be like, decompressed and transformed back into an analog signal type.”(Zhang, Tang & Cai, 2014).

Sound transmission over a packet switched network is further proficient than the sound transmission over a circuit switched network (Casner et.al. 2012). Bandwidth is also saved as only the conversation data is send and the quiet periods are ignored. After removing the quite periods, the savings has increased.

In a network of circuit switch, the whole circuit put away only a single call at one time. (Xu (2014)

In a network of packet switch, digital figures are slice up into small package, send into the system, and forward to the goal (Afshar et.al 2014). This type of circuits only takes the necessary bandwidth in every packet hence much more transmission can made at the same time.

Internet Telephony just takes the benefit of the efficiencies of packet switched networks. IP Telephony is mostly possible due to the gateway (Zhou et.al. 2014).

Gateways are the main element necessary to make possible to the IP telephony. The usual circuit switched PSTN is connected with the internet of packet switch through the gateway (Sheikh Et.al 2014). The gateway converts the input signal into the required signal and moves the call from one network to other.

Requirements

The requirements of VoIP are divided into 3parts:

  • Requirements of Software
  • Requirements of Hardware
  • Requirements of Protocol

Tone dispensation unit: This unit is essential to prepare the tone samples for broadcast. The tone-dispensation unit must maintain the functions like-

A Pulse-code modulation(PCM) Interface is essential to obtain model from the telephony boundary and promote them to the VoIP software for additional dealing out.

Echo Cancellation is essential to decrease or remove the echo, resulted beyond 50 milliseconds (Marshalle et.al. 2013).

Unused Noise recognition is essential to restrain packet broadcast on the network when no tone signals are send (Afshar et.al. 2014). Hence the network traffic is decreased as most of the voice calls are quiet.

A Voice Detector is essential to distinguish among tone and fax indicator (Casner et.al. 2012).

The package tone protocol is essential to summarize compacted tone and fax information for spread through the system.

A part of tone playback is essential at the end to safeguard the arriving package sooner than are send for decompression to the Codec (Layman et.al. 2013).

Call signaling unit is essential to provide as an indicator gateway which recognize the call  to excess the network of packet switch unlike to the network of a circuit switch.

Packet dealing out unit: This unit is essential to progression the voice and signals prepared to spread on the IP support system.

Network administration procedure: permit error accounting and design supervision to be execute.

The accurate hardware necessary is depends on managerial requirements and resources. The most noticeable constraint is the subsistence of an IP support structure surrounded by the area office gateway is necessary to link the difference between the IP based network protocol and the PSTN protocol (Afshar et.al. 2014).

The gateway receive a usual telephone gesture and digitizes it sooner than compress it by Codec (Casner et.al. 2012). The compacted information is place into IP envelope and these envelope are routed over the system to the projected goal. The Full Duplex Voice Cards is also needed, which permit both side to talk simultaneously. A secondary door must be calculated as a support in the incident of the crash of the key gateway.

There is some main protocol such as:

H.323 is an ITU (International Telecommunications Union) standard, through which audio/visual conferencing information transmit through the system. H.323 relies on the RTP (Real- Time Transport Protocol) and RTCP (Real Time Control Protocol) on peak of UDP (User Datagram Protocol) to carry aural stream crossways packet support network (Layman et.al. 2013).

Types of VoIP

G.723.1 defines how an aural gesture with a bandwidth of 3.4 KHz must be programmed for stretch at the rate of 5.3 Kbps and 6.4 Kbps (Marshall Et.al 2013). G.723.1 requires a very short communication rate and transport close to mover class feature. The VoIP discussion as the bottom line Codec for short bit rate IP Telephony has selected this encoding method.

G.711 – The ITU uniform PCM as G.711. This permit the carrier rank quality aural motion to be determined for diffusion at the rate of 56 Kbps or 64 Kbps (Casner Et.al 2012). G.711 uses A-Law amplitude density and is the bottom line constraint for the majority ITU multimedia transportation standard.

Real-Time Transport Protocol (RTP) is a system protocol for distributing aural and video over IP network (Nortz et.al. 2014).

Resource Reservation Protocol (RSVP) is the procedure that chains the condition of resources crossways an IP network.

VoIP has generally three main branches:

VoIP over Internet-This is perhaps the finest identified chatting PC to PC. Mostly telephone calls with no charge. If both the parties have admission to the public Internet at nil cost then the call will be free (Layman et.al. 2013).

Advantage: Free calls do not matter of distance or length of call.

Disadvantage: Call can be available also when the voice quality is low.

Office to office- There is computer to computer access in every multinational business. By set up a VoIP Gateway in every office and linking it to the office heritage Private Branch Exchange (PBX) can provide a voice call among offices.

Advantage: Charges are free for interoffice voice call; voice class is fine, since the company has its own bandwidth.

Disadvantage: There must be an additional bandwidth which recompenses the reserves in the offices.

IP PBX-An established Private Branch exchange (PBX) join all the telephone inside an association to the civic telephone set-up. All the internet phones are replaced to VoIP telephones by IP PBX (Casner et.al. 2012).

Advantage: Infrastructure is distinct cable. The tools are clear to the user, and need lowest amount of guidance.

Disadvantage: Mainly useful for Greenfield sites, but can be modified to work with accessible technology.

Conclusion

VoIP is a technology through which only using a broadband internet connection voice call can be done in place of an ordinary phone line. VoIP services might simply permit to call via the similar service, but others may permit to call any person having a telephone number together with local, long space, mobile phone and global number. Though a few VoIP service only employment in excess of your PC or a particular VoIP telephone, further services permit you to use an established telephone linked to a VoIP adapter.

References

Dickinson, R., Marshall, R., & Helme, S. P. (2015). U.S. Patent No. 20,150,009,900. Washington, DC: U.S. Patent and Trademark Office.

Zhang, L., Tang, S., & Cai, Z. (2014). Efficient and flexible password authenticated key agreement for voice over internet protocol session initiation protocol using smart card. International Journal of communication systems,27(11), 2691-2702.

Jackson, D. C., Epstein, M. J., Giannandrea, J., & Verber, M. A. (2014). U.S. Patent No. 8,705,519. Washington, DC: U.S. Patent and Trademark Office.

Xu, S. (2014). U.S. Patent No. 8,724,547. Washington, DC: U.S. Patent and Trademark Office.

Afshar, S. K., Ezrol, G. M., Fisher, S., Glazer, S., & Roy, R. R. (2014). U.S. Patent No. 8,630,299. Washington, DC: U.S. Patent and Trademark Office.

Zhou, A. H., Zhou, T. T., Zhou, X., & Dylan, T. (2014). U.S. Patent No. 20,140,330,656. Washington, DC: U.S. Patent and Trademark Office.

Sheikh, N. A., Malik, A. A., Mahboob, A., & Nisa, K. (2014). Implementing voice over Internet protocol in mobile ad hoc network–analysing its features regarding efficiency, reliability and security. The Journal of Engineering, 1(1).

Layman, R., Smith, R. M., Meyers, K., & Fleming, B. (2013). U.S. Patent Application 13/831,785.

Zhu, Y., Dickinson, R., Marshall, R., & Helme, S. P. (2013). U.S. Patent No. 8,385,881. Washington, DC: U.S. Patent and Trademark Office.

Casner, S. L., Poduri, K., Alonso, R. G., Alaettinoglu, C., & Jacobson, V. (2012). U.S. Patent No. 8,274,901. Washington, DC: U.S. Patent and Trademark Office.

Dickinson, R., Marshall, R., & Helme, S. P. (2011). U.S. Patent No. 7,903,791. Washington, DC: U.S. Patent and Trademark Office.

Casner, S. L., Poduri, K., Alonso, R. G., Alaettinoglu, C., & Jacobson, V. (2012). U.S. Patent No. 8,274,901. Washington, DC: U.S. Patent and Trademark Office.

Nortz, D. M., Ramakrishnan, K. K., Kalmanek Jr, C. R., Marshall, W. T., & Mishra, P. P. (2014). U.S. Patent Application 14/521,501.

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[Accessed 01 March 2024].

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