Top PDF II.B ACKGROUND A. VoIP (Voice over Internet Protocol)

II.B ACKGROUND A. VoIP (Voice over Internet Protocol)

II.B ACKGROUND A. VoIP (Voice over Internet Protocol)

The contention based IEEE 802.11x Wireless LAN or WLAN is derived from the non-wireless Ethernet 802.3 Local Area Network (LAN) access technology. Layer 1 (physical) and Layer 2 (data link) operate over two frequency bands of 2 GHz and 5 GHz. 802.11b (11 Mbit/s) and 802.11g (54 Mbit/s) are two common standards with a range of between 80-100 m. Contention will reduce the practical bitrates and affect the QoS for real-time traffic, especially VoIP. The large packet headers, for both the WiFi and VoIP, constrain the payload capacity and affect the QoS further in times of congestion. WiFi is cheap and comes as a standard feature on most network devices. WiFi can be found in: public transportation, public spaces, domestic and industrial applications [10].
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Intra Networking (Establishing a Private Branch Exchange)

Intra Networking (Establishing a Private Branch Exchange)

These days most corporations have their own internet which can be accessible only to those within the company. The main aim of an intranet is it often provides a wide range of information and services to an organization that are unavailable to public, unlike the internet. Intranet can be accessed from internet but with restrictions as it is designed to a private space. For example, if in a company it is doubtful that an employee is able to access all the information regarding the company, so that employee can be restricted by giving him only access to certain areas of internet which is relevant to his job. Therefore, to communicate over intranet a PBX (Private Branch Exchange) is supposed to be established which uses the protocol named Voice Over Internet Protocol (VoIP) and an open source communications toolkit which powers IP PBX systems i.e. Asterisk[1]. A PBX is a telephone system within an enterprise that switches calls between enterprise users on local communication lines. VOIP (Voice Over IP) is the protocol that is used by PBX. It helps in transmission of voice and multimedia content over Internet Protocols (IP). This technology allows voice calls over internet instead of the regular telephone lines. In order to establish a PBX server with Ubuntu as the operating system and an open source software named 'Asterisk" for defining the users of an enterprise in the software. A PBX acts as the central switching system for phone calls within an
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Voice over IP Via IEEE 802.11 Wireless LAN

Voice over IP Via IEEE 802.11 Wireless LAN

number of factors are involved in making a high-quality VoIP call. These factors include the speech codec, packetization, packet loss, delay, delay variation, and the network architecture to provide QoS. Other factors involved in making a successful VOIP call include the call setup signaling protocol, call admission control, security concerns, and the ability to traverse NAT and firewall. The application on implementing Sinch-based VoIP (Voice over internet protocol) is for Smartphone Android OS mobile. The purpose of this application is to implement a program that uses VoIP as a means of communication between mobile phones at no cost. The system will allow users to search for other individuals within WIFI range and to establish free connection for voice communication. If the users are not within the WIFI range, they cannot communicate with each other. If the users are not connected with WIFI, they can also communicate with other users by means of mobile data network.
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A FUZZY APPROACH FOR REPRESENTATIVE NODE SELECTION IN CROSS LAYER TCP

A FUZZY APPROACH FOR REPRESENTATIVE NODE SELECTION IN CROSS LAYER TCP

The simplicity of expanding and maintenance wireless network helped to use it widely in the public locations such as parks, airports and bus stations. Distributed Coordination Function (DCF) and Enhanced Distributed Channel Access (EDCA) protocols are used with WLANs. DCF protocol sends its data without any priority between different data types, and does not give any advantage of real time application such as voice over internet protocol (VOIP) or video conference. However EDCA protocol divides data to different class’s voice, video, best effort and background. Each class has its own priority and parameters. The default values of EDCA protocol give voice traffic the highest priority. Quality of Service (QOS) parameters such as end to end delay, packet loss and jitter values determine if the protocol support QOS or otherwise. This paper evaluates the performance of DCF protocol and EDCA protocol by using OPNET simulation and shows the differences between them, depending on QOS parameter values. The result of simulation shows the limitation of EDCA protocol when increasing the number of VOIP users. Therefore the EDCA protocol tolerates specific number of real time applications with acceptable values of QOS parameters.
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Quality of service in IEEE 802 11ac and 802 11n wireless protocols with applications in medical environments

Quality of service in IEEE 802 11ac and 802 11n wireless protocols with applications in medical environments

Abstract: Wireless computer networks are increasingly important as reliable means of communication in medical environments. Evaluation of Quality of Service (QoS) in wireless computer networks deployed in medical environments can improve network performance and enhance utilization of resources. In this study, the QoS offered by IEEE 802.11n and IEEE 802.11ac wireless protocols was evaluated and compared using multiple point-to-point links for Voice Over Internet Protocol (VoIP) traffic. QoS was evaluated based on Predictive Statistical Diagnosis (PSD) and Probabilistic Neural Network (PNN). PSD and PNN based QoS evaluation methods categorized the VoIP packets into low, medium and high QoS types based on the packets' transmission delay, jitter, and percentage packet loss ratio. Both PSD and PNN allowed QoS for VoIP to be quantified accurately. It was shown that 802.11ac provides a higher QoS for VoIP transmission as compared with IEEE 802.11n. The devised methods can be used in medical environments for evaluation of wireless networks' QoS.
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Wireless Enabled Voice over Internet Protocol (VoIP) Network Application Using Asterisk PBX

Wireless Enabled Voice over Internet Protocol (VoIP) Network Application Using Asterisk PBX

prices, there will be no full maturity usage of the internet in Sub Saharan Africa, and more so users in low resource settings. For example, a survey of mobile phone usage in five African countries (Kenya, Uganda, South Africa, Ghana, and Nigeria) has found that Internet browsing via phones now stands at 40 per cent across these countries, with 51% of respondents in Ghana and 47% in Nigeria reporting that they use their phones to access the internet. South Africa lags behind at 40%, and Kenya (34%) and Uganda (29%) are slowest on the uptake (http://www.itnewsafrica.com/). Consumers in South Africa, one of the countries which have a higher increased uptake of smartphones, are at the forefront in voicing their discomfort with the high data prices via #Datamustfall [4]. Comparatively, South Africa’s data prices are much higher than those in other countries on the continent and in the world. This is concerning because according to [5], unless we tackle the issue of unaffordable internet, we're just going to go on entrenching and worsening the existing divides and inequalities in our country`. Consumers in the low remote areas of Sub Saharan Africa, who are unable to afford high costs of data, will be worse off in this digital age. This problem is more severe for students who are in areas with shortages of qualified teachers and traditional
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Performance Analysis Of Active Queue Management (AQM) In VOIP Using Different Voice Encoder Scheme

Performance Analysis Of Active Queue Management (AQM) In VOIP Using Different Voice Encoder Scheme

Despite the growing popularity of data services, voice services still remain the largest income source for network service providers. The two most popular ways of providing voice services are packet switched telephone networks (PSTNs) and wireless cellular networks. The deployment of both of these forms of networks requires infrastructures that are usually very expensive. Alternative solutions are being sought which can deliver good-quality voice services at a relatively lower cost. One way to achieve low cost is to use the already existing IP infra-structure. Means that carry voice with computer data in the same links, Protocols that are used to carry voice signals over the IP network are commonly referred to as voice-over-IP (VoIP) protocols [2]. VoIP application typically works as follows. First, a voice signal is sampled, digitized, and encoded using a given algorithm/coder. The encoded data (called frames) is packetized and transmitted using RTP/UDP/IP. At the receiver’s side, data is de-packetized and forwarded to a play out buffer, which smoothest out the delay incurred in the network. Finally, the data is decoded and the voice signal is reconstructed. A wide range of impairments are taken account such as codec impairments, end to end delay, jitter, and packet loss, as well as noise and echo.in VoIP voice signal is compressed and converted to digital voice packets, VoIP then uses the Internet Protocol (IP) for managing voice packets over IP network. Therefore, VoIP can be deployed on any IP enabled data network, such as the Internet, Ethernet, fabric or wireless network [3]. There are several different types of VoIP service depending on the infrastructure used for the communication: computer to computer based VoIP (VoIP device to another VoIP device), computer to Phone based VoIP (VoIP device to a PSTN device), and Phone to Phone based VoIP (PSTN device to another PSTN device) [4]. Each type of them has __________________________
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Performance Of VoIP In Wireless Multi-Hop Network

Performance Of VoIP In Wireless Multi-Hop Network

In recent years, we have witnessed a growing interest in the transmission of voice using the packet-based protocols. Voice over Internet protocol (VoIP) is a rapidly growing technology that enables the transport of voice over data networks such as the public Internet. The idea of VoIP has been discussed since the early 1970 s when the idea and technology were developed. However, at that time VoIP did not find wide acceptance and deployment both among users and telecommunication providers. This was mainly due to the lack of IP infrastructure and the fact that circuit-switched calling was still a much more reliable alternative, especially in the light of the poor quality of early VoIP calls.
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Modelling And Simulation Of Voice Over Internet Protocol (Voip) Over Wireless Local Area Network (WLAN)

Modelling And Simulation Of Voice Over Internet Protocol (Voip) Over Wireless Local Area Network (WLAN)

Several researches have been done on VoIP deployment since its advent. Though VoIP over WLAN is one of the most recent applications in the high-speed packet-switched networks, nevertheless several essential researches are being carried out to ensure a good Quality of Service. Detailed simulation approach for deploying an end – to end VoIP component from sender to receiver was implemented in [6] and [7] using the OPNET network simulator. The implementation of a practical simulation tool to design and analyze communication networks were described in [1], [2] and [8]. The papers focused on the implementation methods and the applications of the designed simulator supporting VoIP. The innovation of the new model in [2] consists in modeling the user behavior instead of the aggregated traffic. Modeling and aggregation of VoIP streams were investigated in [8]. Development and validation of models for the multiplexed process was presented based on statistical analyses of VoIP traffic in [9], [10], [11] and [12]. The models can be used for simulation of any IP network architecture, wire line or wireless. The analysis in [11] was performed by importing the data derived from the post processing into Matlab software. A theoretical approach to model an aggregated flow of VoIP connections on packet basis created due to tele-traffic parameters were presented in [13], [14] and [15] to evaluate the performance of VoIP services with different Codecs. Simulations were conducted using the one hybrid UMTS in [13] and the Network Simulation 2 in [14] and [15]. In [16], [17], [18] and [19] various mathematical models for measuring VoIP qualities such as Mean Opinion Score (MOS), E-model, and Perpetual Evaluation of Speech Quality (PESQ) score were analyzed. Experimental speech quality measurements under wired and wireless scenarios were compared with the mathematical speech predictor. Findings indicate that WLAN QoS parameters have a high variability in real-world environments, with a significant effect on application performance.
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Security on Voice over Internet Protocol from Spoofing Attacks

Security on Voice over Internet Protocol from Spoofing Attacks

Voice over Internet protocol (VoIP), there is an existing way of communication over any network. Using this technology the users can make telephone calls over IP network. This paper will describe Voice over Internet protocols (VoIP) to a level that allow discussion of security issues and concerns. There are two spoofing attacks are possible, one is IP spoofing attack and another is URI spoofing attack, which are designate in this paper. The implementation of VoIP concerned by businesses, components of a VOIP system, and related security issues. The business apprehensions will be those which are used to affect the Quality of Service (QoS). The network components gateways, call processors and two of the more mutualdesignsapprehended by VoIP.
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Measuring the Success of Adoption of Voice over Internet Protocol; A Case Study of Ministry of Municipality

Measuring the Success of Adoption of Voice over Internet Protocol; A Case Study of Ministry of Municipality

The VOIP technology has one main downside during implementation phase which is the lack of stability. As per the interviewee the new system had major problems such as Quality of the Sound and availability. MUN chose to have the new technology deployed in stages creating a hybrid environment where for a certain time both communication systems were working at the same time to deliver the service. While the implementation phase it was common for the organization VOIP failure due to hardware and software incompatibilities and issues while configuration. This affected number of users not being able to communicate. At some point users had to use 2 types of phones. The CISCO IP phone and the analog phone when the VOIP system fails. This situation caused some confusions for the users specially when the call forwarding service is down and a call needed to be forwarded. The problems got eliminated once the VOIP system is fully implemented and stable, but in fact it left a bad impression. This created a less accepting situation for the new system. However, once the system is fully deployed and stable with no failure events, they started to accept the new technology and created new positive ideas comparing the possibilities VOIP technology presented. Another issue was the phones themselves which depends on the small amount of power signal through the network cable. So, when a power outage occurs then no communication can be done. This leaded the ministry to consider a backup power solution in such cases to avoid this issue. Another issue raised by the interviewee is the security, all the communications must meet a high level of security to secure such private information shared. Especially when citizen’s information is involved. This leads extra testing for the new system and faced by some obstacles (Roysden, R. J., & Schiller, S. Z. ,2015).
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Steganalysis of compressed speech to detect covert voice over Internet protocol channels

Steganalysis of compressed speech to detect covert voice over Internet protocol channels

Figure 4 illustrates the testing process in which stegnography and steganalysis in VoIP streams were carried out over an intranet called CERNET. Alice and Bob communicated secretly, so the VoIP streams between them contain secret messages, which were embedded in the parameters of ITU-T G.723.1 (6.3Kbps) compressed speech streams. Similarly covert communications also occurred between John and Smith. Using our proposed steganalysis method, Mary as a warden monitored the router connected with the network by examining all transmitted streams between Alice, Bob, John and Smith so as to decide whether a transmission contains a hidden message and to estimate the embedding rate.
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Interactive Voice Response System by Using Asterisk

Interactive Voice Response System by Using Asterisk

Voice over IP (VoIP) is one of the widely used protocols for transfer voice data over Internet. More specifically, Voice over Internet Protocol (also called VoIP, IP Telephony, Internet telephony, Network Telephony and Digital Phone) is the routing of voice conversations over the Internet or any other IP-based network. The voice data flows over a general-purpose packet-switched network, instead of traditional dedicated, circuit-switched voice transmission lines. While Voice over IP is often thought as little more than a method of obtaining free long-distance calling, the real value of VoIP is that it allows voice to become nothing more than another application in the data network. Thus allows all internet users to simple use VoIP for communicating each other without any additional charge. On the other hand VoIP does not only offer free long-distance call, but also a number of other services. VoIP makes easy some things that are difficult with traditional phone networks.
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Covert Voice over Internet Protocol communications based on spatial model

Covert Voice over Internet Protocol communications based on spatial model

In the Internet environment, UDP protocol is generally used to transfer streaming media such as VoIP, etc., thus, packet loss is inevitable. And at different periods, because of the different numbers of users on the Internet, there is a great difference in the size of network traffic. Therefore, the packet loss rate varies with time in VoIP communications over the Internet. In our experimental environment, the average packet loss rate was 17% for one week between 1 September 2015 and 6 September 2015; the average annual packet loss rate was about 13% between 1 September 2014 and 1 September 2015, indicating high variable packet loss rates for VoIP communications.
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Characteristic Analysis of VoIP Traffic for Wireless Networks in Comparison with CBR using QualNet Network Simulator

Characteristic Analysis of VoIP Traffic for Wireless Networks in Comparison with CBR using QualNet Network Simulator

Voice over Internet Protocol (VoIP) is one of the most important technologies in the world of communication. Around 20 years of research on VoIP, some Quality of Service (QoS) problems of VoIP are still remaining. During the past decade and with growing of wireless technologies, we have seen that many papers turn their concentration from Wired-LAN to Wireless-LAN. VoIP over Wireless LAN (WLAN) faces many challenges, due to the loose nature of wireless network. Issues like providing QoS at a good level, dedicating capacity for calls and having secure calls is more difficult rather than wired LAN. Therefore VoIP over WLAN (VoWLAN) remains a challenging research topic. Popularity of Voice over IP (VoIP) application such as Google Talk, Skype and MSN Messenger is making VoIP over WiMAX an eye-catching market and a driving force for both carriers and equipment suppliers. In this paper experimentally measured VoIP traffic and CBR traffic over the WiMAX network by using the different routing algorithm using QualNet 4.5.1 network simulation tool. In addition, this paper also shows that VoIP application can be best served with WiMAX rather than the CBR. Future work will involve simulating scenarios that are closer to the real world models.
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Implementation of Voice over IP on WiFi Back Bone

Implementation of Voice over IP on WiFi Back Bone

Voice over internet protocol phones is the wave of future. As the technology improves VoIP phone systems will continue to thrive as a more efficient, cost effective way to terminate phone calls. The main effect of using VoIP is cost efficiency. It is the main use for long distance communication. Before VoIP , internet users expecting the risk associated with sending data over internet. The issue of the VoIP technology is it is not secured. To defeat VoIP security threats, a well secured plan needs to be designed. The plan should include voice encryption, authentication, voice specific firewalls and separation of data and voice traffic. Using intrusion prevention system (IPS) and specific firewalls changing default passwords on different components of VoIP , system used for securing VoIP. VoIP is not traceable as normal telephony services and so will not work with emergency numbers. Steps to rectify this should have sufficient Bandwidth so that information moves quickly. The main things to consider in VoIP are Jitter which is received in wrong order to receiver and can be overcome by using jitter buffer and Latency which is delay in amount of time to reach destination this can be recovered by having quality network management and bandwidth reservation. The Session Initiation Protocol used here to carry VoIP traffic.SIP trunking which means use of voice over IP to facilitate the connection of private branch exchange to internet.SIP trunking with VoIP provides a new foundation for advanced services. SIP holds lot of promise in today and tomorrow communication world. It has been established as a standard for call control and signaling on 3G cell phone networks by the Third Generation Partnership Project (3GPP). This means that all multimedia and IP voice call signaling will be done through SIP. New services involving fixed network IP services can thus introduced with ease.
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Customer management solutions for voice over internet protocol (VOIP) services

Customer management solutions for voice over internet protocol (VOIP) services

This system will benefit VoIP businesses ranging from small to medium enterprises. It will be their main business tools that will enable them to monitor their business and manage their customers and focus on providing reliable service. This system will also allow them to run their business around the clock which will definitely contribute in maximizing profitability.

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Issues and Flaws in Endpoint based Call Admission Control for Voice over WLAN

Issues and Flaws in Endpoint based Call Admission Control for Voice over WLAN

This paper presents the various issues and drawbacks of the popular and existing scenario of carrying the Voice over the WLAN . The existing scenario that manages the Voice over Internet Protocol (VoIP), over an IEEE 802.11 WLAN is the Endpoint based Call Admission Control (CAC). Even though the most researches and experiments over the Endpoint based Call Admission Control shows it to the most effective and yielding acceptable QoS, but still there are certain drawbacks and issues that arises in the real time scenario. Therefore, this paper focuses on those areas of this methodology that leads to flaws and various techniques through which issues can be reduced.
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Measuring The Benefits Of Voice-Over-Internet Protocol (VOIP)

Measuring The Benefits Of Voice-Over-Internet Protocol (VOIP)

It remains important to maintain the ability to trouble shoot your network for potential issues. As a result, companies need to make sure their tools possess the capability to deliver granular diagnostics and reporting abilities. One tool should be able to both identify the problem as well as fix it. Experts recommend using software-based tools as opposed to hardware-based. Software tools can assess and monitor networks from one central location, rather than require on site plug in to the system. This attribute saves time and money. In addition, software updates should support advanced simulation capabilities which facilitate data traffic testing. This benefit functions to determine the maximum acceptable quality of VoIP capacity in the organization. 14
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Efficient and Economic IP Private Branch Exchange for Organization

Efficient and Economic IP Private Branch Exchange for Organization

Internet telephony refers to communications services— Voice, fax, SMS, and/or voice-messaging applications—that are transported via the Internet, rather than the public switched telephone network (PSTN). The steps involved in originating a VoIP telephone call are signaling and media channel setup, digitization of the analog voice signal, encoding, packetization, and transmission as Internet Protocol (IP) packets over a packet-switched network. On the receiving side, similar steps (usually in the reverse order) such as reception of the IP packets, decoding of the packets and digital-to-analog conversion reproduce the original voice stream. Even though IP Telephony and VoIP are terms that are used interchangeably, they are actually different; IP telephony has to do with digital telephony systems that use IP protocols for voice communication while VoIP is actually a subset of IP Telephony. Voice over Internet Protocol (VoIP) is a technology for voice communication that uses the ubiquity of IP-based networks to rely VoIP client devices such as desktop IP phones called Session Initiation Protocol (SIP) phone or soft phone, in an increasing number of businesses and homes around the world because IP is the protocol connecting almost all devices. Voice over IP (VoIP) uses the Internet Protocol (IP) to transmit voice as packets over an IP based network.
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