PDF | Abstract: The Voice Over Internet Protocol (VOIP) is as a combination of IP networks, voice applications and voice calls which being replaced by the old. "What Voice over Internet Protocol (VoIP) is going to do is start to weaken VoIP aspires to be a telephony substitute, it will invite the threat of. and Associated Protocols | VOIP References | Books on Voice over IP and IP Voice over IP (VOIP) uses the Internet Protocol (IP) to transmit voice as http:// ronaldweinland.info~hgs/papers/ronaldweinland.info
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Voice over Internet Protocol (VoIP) is a form of communication that allows you to make phone calls over a broadband internet connection instead of typical. Voice over Internet Protocol (VoIP) is the digital transmission of voice communications through a data network based on Internet. Protocol (IP). PDF | This paper was written for an Independent Study course. Princy Mehta Overview of Voice over IP Professor Udani February 1 Table.
Some VoIP services may only allow you to call other people using the same service, but others may allow you to call anyone who has a telephone number - including local, long distance, mobile, and international numbers. If you are calling a regular phone number, the signal is converted to a regular telephone signal before it reaches the destination. VoIP can allow you to make a call directly from a computer, a special VoIP phone, or a traditional phone connected to a special adapter. In addition, wireless "hot spots" in locations such as airports, parks, and cafes allow you to connect to the Internet and may enable you to use VoIP service wirelessly. A broadband high speed Internet connection is required. This can be through a cable modem, or high speed services such as DSL or a local area network. A computer, adaptor, or specialized phone is required.
Protocols[ edit ] Voice over IP has been implemented in various ways using both proprietary protocols and protocols based on open standards. These protocols can be used by a VoIP phone , special-purpose software, a mobile application or integrated into a web page. Full-service VoIP phone companies provide inbound and outbound service with direct inbound dialing. Many offer unlimited domestic calling and sometimes international calls for a flat monthly subscription fee.
Phone calls between subscribers of the same provider are usually free when flat-fee service is not available. These are typically designed in the style of traditional digital business telephones.
An analog telephone adapter connects to the network and implements the electronics and firmware to operate a conventional analog telephone attached through a modular phone jack. Some residential Internet gateways and cablemodems have this function built in. Softphone application software installed on a networked computer that is equipped with a microphone and speaker, or headset. The application typically presents a dial pad and display field to the user to operate the application by mouse clicks or keyboard input.
VoIP switches may run on commodity hardware, such as personal computers.
Rather than closed architectures, these devices rely on standard interfaces. Dual-mode phones enable users to continue their conversations as they move between an outside cellular service and an internal Wi-Fi network, so that it is no longer necessary to carry both a desktop phone and a cell phone. Maintenance becomes simpler as there are fewer devices to oversee. Two kinds of service providers are operating in this space: one set is focused on VoIP for medium to large enterprises, while another is targeting the small-to-medium business SMB market.
It is a best-effort network without fundamental Quality of Service QoS guarantees. Voice, and all other data, travels in packets over IP networks with fixed maximum capacity. This system may be more prone to data loss in the presence of congestion [a] than traditional circuit switched systems; a circuit switched system of insufficient capacity will refuse new connections while carrying the remainder without impairment, while the quality of real-time data such as telephone conversations on packet-switched networks degrades dramatically.
Fixed delays cannot be controlled as they are caused by the physical distance the packets travel. Latency can be minimized by marking voice packets as being delay-sensitive with QoS methods such as DiffServ. Excessive load on a link can cause congestion and associated queueing delays and packet loss. This signals a transport protocol like TCP to reduce its transmission rate to alleviate the congestion.
VoIP endpoints usually have to wait for completion of transmission of previous packets before new data may be sent. Although it is possible to preempt abort a less important packet in mid-transmission, this is not commonly done, especially on high-speed links where transmission times are short even for maximum-sized packets.
But since every packet must contain protocol headers, this increases relative header overhead on every link traversed. Packet delay variation results from changes in queuing delay along a given network path due to competition from other users for the same transmission links.
VoIP receivers accommodate this variation by storing incoming packets briefly in a playout buffer , deliberately increasing latency to improve the chance that each packet will be on hand when it is time for the voice engine to play it.
The added delay is thus a compromise between excessive latency and excessive dropout , i. Although jitter is a random variable, it is the sum of several other random variables which are at least somewhat independent: the individual queuing delays of the routers along the Internet path in question. Motivated by the central limit theorem , jitter can be modeled as a gaussian random variable. This suggests continually estimating the mean delay and its standard deviation and setting the playout delay so that only packets delayed more than several standard deviations above the mean will arrive too late to be useful.
In practice, the variance in latency of many Internet paths is dominated by a small number often one of relatively slow and congested bottleneck links. Most Internet backbone links are now so fast e. RFC VoIP metrics reports are intended to support real time feedback related to QoS problems, the exchange of information between the endpoints for improved call quality calculation and a variety of other applications.
This is generally down to the poor access to superfast broadband in rural country areas. With the release of 4G data, there is a potential for corporate users based outside of populated areas to switch their internet connection to 4G data, which is comparatively as fast as a regular superfast broadband connection.
This greatly enhances the overall quality and user experience of a VoIP system in these areas. A virtual circuit identifier VCI is part of the 5-byte header on every ATM cell, so the transmitter can multiplex the active virtual circuits VCs in any arbitrary order.
Cells from the same VC are always sent sequentially. Every Ethernet frame must be completely transmitted before another can begin. If a second VC were established, given high priority and reserved for VoIP, then a low priority data packet could be suspended in mid-transmission and a VoIP packet sent right away on the high priority VC. Then the link would pick up the low priority VC where it left off.
VoIP can allow you to make a call directly from a computer, a special VoIP phone, or a traditional phone connected to a special adapter. In addition, wireless "hot spots" in locations such as airports, parks, and cafes allow you to connect to the Internet and may enable you to use VoIP service wirelessly.
A broadband high speed Internet connection is required. This can be through a cable modem, or high speed services such as DSL or a local area network. A computer, adaptor, or specialized phone is required. If you use your computer, you will need some software and an inexpensive microphone. Special VoIP phones plug directly into your broadband connection and operate largely like a traditional telephone.
If you use a telephone with a VoIP adapter, you'll be able to dial just as you always have, and the service provider may also provide a dial tone.
Some VoIP providers offer their services for free, normally only for calls to other subscribers to the service. Your VoIP provider may permit you to select an area code different from the area in which you live.
It also means that people who call you may incur long distance charges depending on their area code and service. Some VoIP providers charge for a long distance call to a number outside your calling area, similar to existing, traditional wireline telephone service. Access Limited Open architecture with almost no restriction Emergency Caller may be localized during No built emergency mechanism.
Security Reasonably secured Unsecured due to its open nature Cost High due to additional Relatively low as existing data network can be infrastructure and management used for transmitting voice It can be observed from the table above that the VoIP Integrity and Availability CIA when this technology is technology is still undergoing some improvement especially being put to use.
This two major factor has been looked into, in terms of quality of service and security. Different QoS with the major improvement on security being the encryption method has been developed and implemented that will of voice packet over both public and private network. In other to isolate attacks, voice and VoIP technology has received diverse reactions and response must be physically secured.
For data traffic must be separated using virtual local those that embraced it, some do not have any concrete area network VLAN. The VoIP traffic must be regulations guiding its deployment and usage. The Regulation having a different physical network. February revealed its planed approach in regulating VoIP . This was followed by series of consultative forum separate server for voice and data network in the and workshops held by the industry study group on VoIP and case of Domain Name Server DNS and Dynamic its impact on international gateways and international access.
Hardening of applications and operating guidelines were announced for Voice over IP regulatory on one of the two networks. Unauthorized traffic must be filter system is essentials so as to protect the network. The network must therefore allow only continue to regulate the service and not the expected traffics on individual VLANs screening out technology.
This enables the industries practitioners unnecessary ones. Consequently, the deployment of VoIP in Nigeria The flow chat presented in figure 2 below illustrates the will be controlled by the commission and the management of VoIP traffic during transmission in a necessary equipment needed for its deployment will secured network.
This means several security solutions applicable to traditional telephony and data network in their present form are not applicable to VoIP  Intrusion detector systems IDS , firewalls and other component must be designed specifically for VoIP so as to emulate and enjoy the security level being enjoyed by PSTN without affecting the voice quality of service. Due to the fact that UDP packet does not guarantee service delivery, network components must prioritize voice traffic over data.
VoIP components must be dedicated as regards to its performance and security. All unused ports must be disabled and the hardware components physically secured. Because it can run security on VoIP. Asterisk runs on Linux deployment by recommending some specially made security platform and was released under GNU General Public solutions for voice traffic consisting of an inbuilt QoS Licence. Various version of application and providing a reasonable amount bandwidth Asterisk has been developed after it was released in that will be sufficient for the intended applications.
It has more than notable new features which include new 5. In our implementation scenario, VoIP will be implemented between two offices in different location. The When designing the VoIP technology to be introduced into Lagos office which is the Head office and the Abuja office the network, the right requirements should be specified so branch office. Among major In this report, an open source Asterisk will be implemented questions identified by  that must be asked are; on a converged network but before then, an assessment will be made on the existing network.
How many users will the IPBX support? Do you have high performance server? What type of operating system will your server run on? With this assessment, potential PSTN trunk line to the system? The Table 2: IP Addressing and Telephone Extension manual editing was used for this implementation.
In this file the various sip clients were defined in IP address This file defines how calls are handled and Voicemail. This file defines an interaction interface with routed within and outside the Asterisk box. The dial plan for callers when the intended receiver is not available to receive various users is also defined in their context that determines the call.
To check the status of the configured users, its extension number, IP address, port number and status, the command used is sip show peers; this is illustrated in figure 3 Figure 3: The call was initiated and received successfully.
Figure 4: Dhamankar, R. Telecommunications globally; therefore integrating it in a converged network in Nigeria must be encouraged. The VoIP .