SIP Security Mechanism Techniques on Voice over Internet Protocol (VoIP) System

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IJCSMC, Vol. 4, Issue. 5, May 2015, pg.424 – 431 RESEARCH ARTICLE A Proposed Security Framework for VoIP

IJCSMC, Vol. 4, Issue. 5, May 2015, pg.424 – 431 RESEARCH ARTICLE A Proposed Security Framework for VoIP

Abstract: VoIP stands for Voice over Internet Protocol and is a way to carry voice traffic over computer networks like the Internet. Over the last decade VoIP has become increasingly popular, gaining millions of subscribers every year (e.g. LINE and WECHAT provide voicecall facilities) and has certainly caught the eye of telecommunication service providers all over the world. The driving factor for the success of VoIP is cost reduction, both for users and providers. But VoIP doesn’t only bring reduced costs it also brings threats and vulnerabilities since it is IP based it’s susceptible to large number of threats. The threats include spoofing or identity theft and call redirection, making data integrity a major risk. Therefore authentication and encryption techniques can be used to design a framework which can survive the possible threats. In this security framework authentication is implemented first to authenticate the true user and then cryptography techniques is used to safely transmit the information stream over the network. The authentication part will be implemented using biometrics because it is not possible to theft anyone’s physical features.
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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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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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Wireless Enabled Voice over Internet Protocol (VoIP) Network Application Using Asterisk PBX

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

Several solutions have been proposed to address student access to the internet within their limited locale such as a campus, but few focus on the context of Sub Saharan Africa. One example is the VillageCell solution by [18] of multiple base stations which are connected through the local wireless network, and calls are routed via private branch exchange (PBX) servers implemented in an open source framework (Asterisk). VillageCell in Africa allows free calls within the local network and standard connections to outside callers using the satellite link. It also uses free, open source solutions and off-the- shelf hardware, and hence the total deployment cost is minimal and the solution is scalable. Examples that feature specifically in the context of education campus include the works of [19] who designed and implemented an Asterisk server to serve as the local exchange for placing voice and video calls within a private WIFI cloud and legacy networks (Public Switch Telephony Network). Their model which is accessible within the area of a university campus, allows only those mobile phones and PCs to connect to the Asterisk server which are registered with the wireless network. [20] Developed a VOIP based Asterisk PBX on Ubuntu 11.10 which allow unlimited free calls with the help of IP phone or traditional phone within a LAN/ WAN without internet connection. For wireless communication, the authors whose study was situated in Asian context where there is limited bandwidth, extended VOIP based PBX to WIFI technology. [21] Present a VOIP based solution designed for the use over a campus environment whose operating costs of phone systems take a large part of the budget. Their proposed solution is designed to carry all telephony signals by offering a new communication tool in which data packet streams are transmitted over IP. Like a standard SIP server, the solution is an IP software which can simultaneously act as a phone gateway for the phone system.
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Review on VOIP over MANET by Utilizing SIP

Review on VOIP over MANET by Utilizing SIP

Voice over internet protocol is an innovation that permits passing voice signal through packet switched lines as opposed to a simple telephone line or analog phone line.[1] It is notable that VOIP applications have inflexible QOS necessities for a steady and reliable communication. The QOS on VOIP is particularly delicate to end to end and packet loss rate. As well as, the kind of voice codec has the effect on the QOS on VOIP because it’s initial function of performing analog/digital voice transferred and digital compact.[14][2] In underground mines, Ad hoc networks are special class of mobile ad hoc network provision real- time information. As of late, the voice over internet protocol system depend upon on session initiation protocol (SIP) has grown quickly.
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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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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

Abstract: Voice over Internet Protocol (VoIP) is a rapidly growing technology that enables transport of voice over data networks such as Ethernet Wide area networks (WANs) due to this important different codec scheme is developed to meet the QoS requirements. This thesis presents a comprehensive study about the impact of active queue management (AQM) on Voice over Internet Protocol (VoIP) quality of service using different codec scheme such as G711, G723, G729 and GSM using simulations tools. The evaluation is done using the OPNET Modeler, which provides a convenient and easy-to- use platform for simulating large scale networks and this also give a power to go through different levels of designing a network even with the ability to program the mechanism you want which is used here to implement two types of AQM mechanism which is not included by default in the OPNET and these two mechanisms are ARED and GRED. The performance metrics used in the study are jitter, throughput and delay. The study shows that G.711 and G729 codecs in a simulation gives a significant result for the performance of VoIP that codec G711 and G.729A has acceptable throughput and less deviation of received to transmit packet as compared to GSM and G.723 also average delay like end to end delay and Voice jitter is lesser in codec G711 and G.729 as compared to the other two referenced codecs.
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Voice-Over-Internet Protocol (VOIP) Cost Efficiencies And The Decision To Implement

Voice-Over-Internet Protocol (VOIP) Cost Efficiencies And The Decision To Implement

Additional savings stems from the elimination of service lines as well as administrative repair and maintenance. Most businesses with 20 or more lines on a PBX network lease a T1 line for their voice network, as well as a separate T1 line for Internet access. VoIP eliminates the need for the second T1 line which can significantly reduce costs. 6 In addition, administrative expenses are cut allowing IT departments to focus time and money on other advancements. IT staff can manage a single network infrastructure from one data center as opposed to maintaining two or three different networks. 11 Since the connection is tied to the individual and not the circuit, companies no longer have to commit maintenance costs when a new employee is added or an existing one transfers offices. The end result is a virtual PBX system without local hardware investment or maintenance. 6
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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)

the computer and the access point, the poorer the signal and the slower the connection, hence large offices often deploy several access points with overlapping ranges. VoIP protocol defines the mechanisms that users in a communication session must follow, depending on the service the end user requires. The protocols are meant to establish end-to-end voice call [20, 13]. From VoIP perspectives, there are two types of protocols; Signaling Protocols and Media Transport Protocols. Signaling protocols provides session setup, control and teardown allowing call information to be carried across network boundaries. Auxiliary function related to setting up and maintaining calls are performed with these protocols [20]. The two most popular signaling protocols are SIP and H.323. Media Transport Protocol is the second type of protocol used in VoIP in the transportation of voice. The protocol is responsible for actually carrying the digitized voice in the form of packets. It ensures the transfer of voice packets through the network. The most popular transport protocols are Real-time Transport Protocol (RTP) and Real- time Control Protocol (RTCP). However, for VoIP over WLAN, the IEEE802.11 Medium Access Control (MAC) protocol defines the method for communication over wireless stations [4, 17]. The MAC protocol ensures that multiple stations can transmit data on the same wireless channel without causing any interference. There are two modes for accessing the shared wireless channel for transmitting data. Distributed Coordination Function (DCF), based on Carrier Sense Multiple Access with Collision Avoidance (CSMA/CA) the first mode is a distributed protocol requiring no centralized entity for coordinating the channel access. The second mode Point
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Customer management solutions for voice over internet protocol (VOIP) services

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

VoIP comprises of two major parts; technical part and management part. Technical part involves tasks such as setting up servers, managing servers and setting up networks for VoIP implementation. The second part involves tasks such as customer management, credit management, and Call Data Record management, and this is where this project comes in. It aims to solve the problem faced by very- technical-based companies that are able to setup their own VoIP service, but has no expertise in developing a system to manage the service. It also targets small companies who are embarking in the VoIP business through a big carrier or
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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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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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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

A number of studies reported QoS evaluation in multimedia wireless networks based on fuzzy logic and neural networks [12][13]. Neural networks were used to analyze and evaluate overall QoS for Voice over Internet Protocol (VoIP) traffic using networks simulated in a package called NS-2 [12]. The traffic parameters were initially classified into three types of QoS by an unsupervised learning Kohonen neural network. The results were then further processed by a supervised learning neural network called multilayer perceptron (MLP) to determine the overall QoS. The QoS measure obtained by the approach was compared with other QoS evaluation techniques that used fuzzy logic and regression modelling [12]. Fuzzy c-means clustering and Kohonen neural network were compared for their effectiveness to classify QoS into high, medium and low types for real time VoIP traffic [13]. The study also used fuzzy inference system (FIS), multilayer perceptron (MLP) and linear regression to quantify overall QoS for real time applications such as VoIP. However, these studies did not evaluate multiple Wi-Fi protocols such as IEEE 802.11ac and 802.11n. QoS assessments methods reported in studies [12][13] were based on either fuzzy c-means, Kohonen neural network, fuzzy inference system (FIS) or MLP. These methods have a number of shortcomings. For example, the FIS approach requires the user to develop the rules for the method's knowledge base and determine the types and parameters of the membership functions for its inputs and outputs. The rules and membership functions' parameters are specific to each particular application. MLP and Kohonen neural networks need careful training to ensure effective generalization [12]. MLP also requires a careful determination of the number of neurons in its hidden layer(s) to avoid overfitting or poor generalization. Kohonen output is a grid of neurons' (map) that requires interpretation to determine the boundary between individual clusters. The performance evaluations carried out in some studies such as [9][14] were not aimed at measuring overall QoS classification and its variation to traffic change.
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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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Securing the SIP Communications with XML Security Mechanisms in VoIP Application

Securing the SIP Communications with XML Security Mechanisms in VoIP Application

The SIP protocol (Initiation Session Protocol) is a protocol of signaling to textual structure allows to establish a connection of communication through computer systems between two or several clients by different means (Email, telephone, Web,…) by using the client-server model. SIP employs procedure DNS that allows a client to solve URI SIP in an address IP or port or a protocol of transportation to an other destination to consult [1,2]. SIP message can be encapsulate on different protocol such TCP and UDP. SIP is described as a protocol of control of the application layer. It establishes, modifies and ends conversations. It gathers in syntax of typical HTTP and SMTP (Simple Mail Transfer Protocol) because it allows the establishment of a session between speakers identified by addresses. The SIP transaction demonstrates in the case where a terminal INVITE a second terminal to participate in a new session (modeled by the syntax INVITE). The request emit to a client, who will seek beside a database the information concerning the second terminal. Also SIP Can open canals of communications by textual process or with the help of the voice as it shows the syntax following:
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A Comprehensive Survey on Voice over Internet Protocol (VoIP)

A Comprehensive Survey on Voice over Internet Protocol (VoIP)

Abstract— Voice over Internet protocol (VoIP) is a new way of communicating. It is a technology that allows users to make telephone calls over an IP network. This paper will describe Voice over Internet Protocol (VoIP) to a level that allows business concerns of implementing VoIP, components of a VoIP system. The business concerns will be those that affect Quality of Service (QoS). VoIP components will include end-user equipment, network components, call processors, gateways and two of the more common architectures: Session Initiation Protocol (SIP), This paper gives a brief introduction of VoIP technology: the network structure, protocols, echo and delay, jitter, and packet loss in VoIP network. Finally, the survey concludes with a discussion on the feasibility of providing VoIP over challenging satellite links.
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A Detail Review on Voice over Internet Protocol (VoIP)

A Detail Review on Voice over Internet Protocol (VoIP)

Abstract— Voice over Internet protocol (VoIP) is a new way of communicating. It is a technology that allows users to make telephone calls over an IP network. This paper will describe Voice over Internet Protocol (VoIP) to a level that allows business concerns of implementing VoIP, components of a VoIP system. The business concerns will be those that affect Quality of Service (QoS). VoIP components will include end-user equipment, network components, call processors, gateways and two of the more common architectures: Session Initiation Protocol (SIP), This paper gives a brief introduction of VoIP technology: the network structure, protocols, echo and delay, jitter, and packet loss in VoIP network. Finally, the survey concludes with a discussion on the feasibility of providing VoIP over challenging satellite links.
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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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II.B ACKGROUND A. VoIP (Voice over Internet Protocol)

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

The entire call setup procedure from the beginning to the end requires the exchange of the signaling and control information between the parties involved. This already complicated process becomes even more so where mobility is involved through a heterogeneous network. Different devices have different capabilities and the seamless flow of information between them needs a priori information about their capabilities before the full flow of information can take place. This is handled by the SIP [5] application layer control protocol working alongside the existing other protocols. The destination and source “user agents” discover each other and establish the seamless connection between them based on shared properties using SIP. SIP creates any necessary proxy servers where needed by intermediary nodes dealing with such events as: registration, invitation and other requests. Mobility features are also catered for including: name mapping; redirection and the maintenance of an external visible location invariant identifier [6]. SIP is both session and device independent handling a wide range of multimedia data exchange capabilities. SIP also has many more functionalities, such as: the ability to setup a multicast call; removal of participants; user availability, session setup, user capabilities, user location and session management.
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Analysis of DDoS Attacks in Heterogeneous VoIP Networks: A Survey

Analysis of DDoS Attacks in Heterogeneous VoIP Networks: A Survey

Safoine R., Mounir S. and Farchi A. [18] presented a study of different flooding detecting mechanisms. Authors compared two particular proposed techniques for anomaly detection and notifying users about possible intrusions. One of those detection algorithms provides solution according to the port number used to transmit the message. It was developed in C language and composed of three processes: collection of traffic from network, analysis of packet headers and detection of SIP based VoIP packets which are unprotected. The other one is a probabilistic solution with two phases: training phase that collects and builds standard VoIP SIP behavior profile based on Poisson distribution and testing phase that detects flooding attacks by comparing current SIP operation profile with the normal profiles built during training phase. Authors proposed a method for the detection of flooding attacks in VoIP networks which provides specific security algorithm depending on the characteristics of the ports used in the transmission. System raises an alarm whenever a packet is considered to be unprotected or as an attack.
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