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Voice over Internet Protocol (VoIP) Technology

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Any opinions, findings, conclusions or recommendations expressed in this material are those of the authors and do not necessarily reflect the views of UK Essays.

Published: Fri, 02 Mar 2018

ABSTRACT:

Voice over Internet Protocol (VoIP) technology which attract extra attention and awareness to the world wide business. IP telephony’s system will affect the sending voice transmission in the form of packet over the IP network in the VoIP applications. Now a day’s many industries will use the VoIP technologies to provide the Security. In this project, I provided a variety of VoIP safety intimidation and probable approach to handle the intimidation in VoIP application.

VoIP is naturally susceptible to networks attack, like hateful codes (i.e., worms, viruses, Trojans), denial-of-service (DoS), distributed DoS (DDoS), pharming, and (though non malicious) sparkle were crowded. These attacks also spoil grimy system by overriding assets, distracting valid user, compromise private informations, or by demeaning code and records. This break affect the contaminated system, it also destroy the unaffected (or even non-vulnerable) ones. All system associated to the Internet are responsive to hateful code which try to contaminate as much as hosts is probable, cause overcrowding on the network communications.

The QoS which is provide to the end user is of highest significance and it is the main issue to implement the VoIP system, since if the exchange is indecipherable then there is no position in giving the service. The major factor which distresses the superiority of services is Latency, Jitter and Packet loss.

CHAPTER-1

1.0 Introduction

The voice enterprises continuously providing voice conversation services on over broad band by discovering the current market issues and network issues from past 20 years, the voice transition industry undergone various security and network issues to produce better quality voice service to transit on over broad band. The current voice market has been step up into the new level of voice protocols for providing VoIP services during low bandwidth, high level of data and voice transmission provisions.

The VOIP technologies allow sharing the resource of WAN for supporting data and voice for saving the cost for transmission process [1]. The VoIP provides many advantages to the enterprises, the migration of voice and telephone application form TDM switch network to IP packet switched network provides many advantages to enterprises during the migration of voice into IP application the enterprises need to provide security to the data applications [2].

However the enterprises need to identify the security issues and employee new techniques to protect against attacks. Security and QoS is a main aspect of VOIP system, the data on voice networks have been attacked by viruses, worms, DOS attacks and other unknown authentication users [3]. The VOIP architecture is a complete network hierarchical structure which is compound with many of the networking devices, the design structure have to ensure that whether the components will cope the unwanted attacks. The protocol structure in a VOIP system is a more sensible factor due to the poor ambitious. Here we introduce the attacks on over VoIP system.

VOIP Overview

VoIP stands for Voice over Internet Protocol which is the mainly used in the transmission of voice communications through IP network like internet, public switched networks [4]. The concept of VoIP mainly targets of the transmission of voice based messages and applications by using different protocols and is transmitted via the internet.

The basic steps involved in the transmission of voice signals through the internet are:

· Conversion of voice to analog and digital signal.

· Compression and conversion of the signal into Internet Protocol Packets to broadcast over Internet.

VoIP systems adopt different session control protocols for commanding over the set-up, tear-down of calls and also different audio codec’s which allow for encoding the voice signal and allow the transmission. These audio codec’s may vary form system to system where some of them are based on the narrow band and some on the compressed speech where some other system may use high fidelity audio codec’s.

Technologies used to implement VoIP:

· H.323 [12]

· IP Multimedia Subsystem (IMS)

· Session initiation Protocol (SIP) [5]

· Real-time Transport Protocol (RTP) [5]

1.1 Problem Definition

In the past days the VoIP security is a not a big concern the people were mainly concerned with the functionalities, cost and the usage, but the VOIP communication trend has been encouraged; the VOIP communication system widely accepted by the people; due to the high acceptance of VOIP system the security issues are main concern.

However the VoIP services are rapidly growing in the current voice communication system, many unauthenticated users and hackers are stealing the VoIP services and hacking the services from the service providers and re routing to their personal usage. Some of the security standards are not credential they only supports to authentication over calls, but the problem with the service theft.

The security concerns will affect on quality of the system, due to the security tools and security solutions will conflict on quality of service. The system will accept the security tools those tools shouldn’t decrease the quality. The basic issue of the quality is firewall. The firewall will blocks the calls for security constrains it will not process the signaling which are allocated to the UDP ports. Due to the security issues on VoIP devices will consumes extra time for packet delivery and which consumes extra time during the call; so it may delay the packet delivery, due to the encryption and decryption mechanism will conflict the call time.

1.2 Objectives of the study

The basic objective of this is to detect source of attacked packet on over network

Ø To formally define the network security problems and unauthorized access incidents

Ø To define the most accredited security techniques and security methods

Ø To evaluate the prototype system and packet feature mechanism

Ø Email and other internet message are easily integrated with the voice applications

Ø To support the multimedia applications, which provides less cost effective services for video conference, gaming

Ø To supports a low cost, flat rate pricing on the voice communication over the

Public Internet and Intranet services.

Ø Sends the call signaling messages over the IP-based data Network with a suitable quality of service and much superior cost benefit.

Ø Present offline message passing between the users by selecting a user from predefined offline user list

Ø Present textual communication

1.3 Research Method

Ø Provide authentication to the end users for accessing the VoIP services

Ø Design secure VoIP Configuration system

Ø Attempt to separate VoIP traffic from normal data traffic using either VLANs or a completely separate physical network.

Ø Enable authentication on SIP accounts.Internal Firewalls/ACLs should be cond to block telnet and http traffic from reaching voice VLANs or subnets.

1.4 SCOPE

These researches analyze the security and performance issues, it has to research on different security levels and represent various security challenges to modern VoIP system.

Ø This research enhance security methods by analyzing the modern security challenges

Ø To present various security methods; this security methods are explained in chapter -3 to analyze and investigate the security threats and define the solution for obtaining better performance

Ø Balance VoIP security and performance by measuring the services and network traffic

Ø To present VoIP protocols for secure data transmission

1,5 Thesis Organization

Chatper-1: Introduction: General Introduction of VoIP, problem definition and Research methods

Chapter -2: Literature Review: Review of VoIP deployment and review of security issues and performance and VoIP security background and security challenges

Chapter -3: Security process: VoIP security process, managing of VoIP security and security process and define the security solutions

Chapter -4: VOIP security and performance: Demonstrate VoIP performance , balancing of security and performance of VoIP

Chapter -5: Analysis Report: security and performance analysis and investigation reports of VoIP security and performance and complete project report scenario

Chapter -6: Conclusion, Future Enhancement, References and Appendices.

CHAPTER -2

2.0 LITERATURE REVIEW

Background

VoIP is a IP telephony which is used to deliver a voice on over internet; which stands for Voice over Internet Protocol which converts a voice signals to digital voice packets and transmit these packets on over network; for transmitting which uses Internet protocol for coordinating voice packets. VoIP can be deployed in dissimilar kind of IP enabled network like Internet, wireless networks, Ethernet.

VoIP is a telephony system which takes voice as a analog signals and which converts it into digital format and transmit on over network by using Intern protocol.

VoIP service Types

VoIP provides different types of voice service according to the communication media infrastructure; the most common services are as follows

Ø Computer to computer based services

Ø PC to phone and phone to PC based services

Ø Phone to phone based VoIP services [6]

Computer to computer:

A voice exchange in between system to system is one type of communication provides free VoIP’s services which it requires related software applications such as gtalk[8], skype[7], messengers. In this services the users need to install same software’s in their respective PC’s and exchange their voices same as Peer to Peer services.

PC to phone and phone to PC:

It is a combination of Internet and circuit switched telephone system. The VoIP application software receives the voice and hand over to the Internet protocol to communicate on over telephone network. VoIP services provide a services to communicate with phone s by establishing VoIP network; an applications such as Skype, messengers are communicate to the phones by converting respective receiving and transmitting formats.

In the Phone to PC services the user can communicate from phones to PC’s; user can dial to PC’s by calling like normal phones; in this services the PC IP address contains a phone number. The user can dial from phone to assigned PC IP address phone number; Skype is a best example for this kind of services, which allows users to purchase a VoIP services to communicate from phone to PC [7].

The most common devices in these services are

Ø VoIP service providers

Ø Modem

Ø Internet services

Ø ATA: Analog Terminal Adaptor, this convert analog signals to voice signals & voice signals to analogs singles

Phone to phone based VoIP services [6]: Now a day’s this type of services are using in long distance calls; many communication service provide companies offering long distance calls in very abnormal price by utilizing the PSTN services.

VoIP System

A Fig- 1 shows a typical VoIP network topology which is a combination of given equipments; the following equipments are

1) Gatekeeper

2) VoIP Gateway

3) VoIP Clients

Gatekeeper: A VoIP gatekeeper is a routing manager and central manager in a H 323 IP telephony surroundings. This is an option in a VoIP system which manages end points of a sector. VoIP gatekeeper is useful for managing calls, terminals and gateways. VoIP gatekeeper presents access control, bandwidth control and address translation.

VoIP gateway:

The VoIP entry convert a voice calls into genuine instant in between Public switch Telephone Network (PSTN) and IP networks. The basic functionalities of VoIP entry are compression, decompression; signal controlling, packetization and call routing.

VoIP clients: This equipment represents phones, multimedia PC’s

2.1 Security Issues.

VoIP Phishing – How To prevent VoIP Phishing and avoided getting Trapped

You can do prevent VoIP Phishing at home and in your corporation and to avoid yourself and your associates from being keen as a Phishing victim.

What is VoIP Phishing and hoe it work

VoIP Phishing is a type of assault that lures the user into given personal data like phone number, credit card numbers, and password over a web site. Phishing over VoIP is become uncontrolled as VoIP makes Phishing easers for attacker.

Security thread in VoIP

While VoIP has become a one of the conventional communication technologies, VoIP user face a serious of security threads let’s see this security issues.

Firewall

A firewall is software is planned to protect a personal networks from illegal access. Firewalls usually block the worthless passage from the outside to the inside of the networks and so on.

Over look security

You must not look at only at the light side of VoIP. While it is revolutionizing voice and data communication, it does not symbolize some problematic security issues that need that need to be deal with accurately.

Quality of Service Issues (Qos)

Qos [9] is a basic process of VoIP; if it delivers a good quality of services to the users which are more advantage to the users for saving money; rather than spending much money on other communication services. The Quality is an importance factor for VoIP services providers industries. In Certain level the security issues implementation can degrade the QoS. The security procedures such as firewalls and encryption techniques block the calls and delay the packet delivery.

The main QoS issues are

Ø Latency

Ø Jitter

Ø Packet loss

Ø Bandwidth problem

Latency:

Latency represents a delivery time for voice transmission from source to destination. The ITU-T advice that G.114 [10] establish a many time of constraints on one-way latency .To achieve Quality of Service the VoIP calls must be achieve in a limited bound time.

The basic issues in latency are

Ø Time spent on routers and long network distance

Ø Security measures

Ø Voice data encoding

Ø Queuing

Ø Packetization

Ø Composition and decomposition

Ø Decoding

Jitter:

The non-uniform packets make a packet delivery delay; which it is caused by insufficient bandwidth. The packets are in out of sequence order, for transmitting voice media it uses RTP protocol; this protocol are based on UDP so that it makes the packet in out of order sequence which degrades the QoS by not resembling the protocols at protocol level.

Packet Loss:

The packet loss increase the latency and jitter; where group of packets are arrived late will be discarded and allow new packets. The packet loss is associated with data network; due to the low bandwidth and high traffic which delays the packet delivery.

Bandwidth:

The low bandwidth delays a packet delivery which degrades the QoS by increasing the latency and jitter. The data on over network have to distribute into various nodes; the data have to transmit from one node to another node during this transmission if it encounter any problem which it can delays the packet.

The entire network design includes routers, firewall and other security measures. Certain time in the network path some of the nodes are unavailable at that time it doesn’t deliver the packets to an end users.

2.2 VoIP protocols

There are numbers and numbers of network that can be working in organize to offer for VoIP communiqué service .In this part we will center no which the general to the best part of device deploy.

Almost each machine in the globe use a standardization called real time protocol (RTP) for transmit of audio and video packet between the networks. IETF is the founder of RPT. The consignment layout of numbers CODE are define in RFC 3551 (The section “RTP profiles and pay load format specification” of RCF. These sections address items.). Though pay load format section are define in document also published by the ITU (International telecommunication union) and in others IETF RFCs. The RTP mostly deal with issue like packets order and give mechanism to help the address wait.

The H.323 [7] standard uses the Internet Engineering Task Force (IETF) RTP protocol to transport media between endpoints. Because of this, H.323 has the same issues as SIP when dealing with network topologies involving NAT. The easiest method is to simply forward the appropriate ports through your NAT device to the internal client. To receive calls, you will always need to forward TCP port 1720 to the client. In addition, you will need to forward the UDP ports for the RTP media and RTCP con-trol streams (see the manual for your device for the port range it requires). Older cli-ents, such as MS Netmeeting, will also require TCP ports forwarded for H.245tunneling (again, see your client’s manual for the port number range). If you have a number of clients behind the NAT device, you will need to use a gate-keeper running in proxy mode. The gatekeeper will require an interface attached to the private IP subnet and the public Internet. Your H.323 client on the private IP subnet will then register to the gatekeeper, which will proxy calls on the clients’ behalf. Note that any external clients that wish to call you will also be required to register with the proxy server. At this time, Asterisk can’t act as an H.323 gatekeeper. You’ll have to use a separate application, such as the open source OpenH323 Gatekeeper

H.323 and SIP Have their origins in 1995 as researchers looked to solve the problem of how to computers can indicate communication in order to exchange audio video files.H.323[12] enjoy the first commercial success due to this fact those who are working on the protocol in ITU[12] worked quickly to publish the first standard in the year 1996.

While support of the two protocols on a single gateway is critical, another integral part of dual-protocol deployment is the ability for H.323 gatekeepers and SIP proxies to interwork and share routing capabilities. One method that was introduced to support time-to-market requirements uses routing interaction between a Cisco SIP Proxy Server and an H.323 gatekeeper.

The business model for some carriers using the Cisco Global Long Distance Solution is to provide origination and termination of voice-over-IP (VoIP) minutes for several other service providers. This business model has been very successful with deployment of H.323-based services, but these Cisco customers would also like to attract additional SIP-based service providers. Ideally, these customers would like to use their existing voice-gateway infrastructure to support additional SIP-based offerings.

Cisco has provided these carriers with a way to add new SIP services by adding capabilities to the Cisco SIP Proxy Server to allow it to “handshake” with an H.323 gatekeeper using the H.323 RAS protocol. By enabling a SIP proxy server to communicate with an H.323 gatekeeper using RAS location request, location confirmation, and location reject messages and responses, a Cisco SIP Proxy Server can obtain optimized routing information from VoIP gateways that have been deployed in the service provider’s network.

The Cisco architecture allows for protocol exibility and enables, one call-by-call basis, use of a particular session protocol. This exibility allows customers to deploy SIP networks on proven packet telephony infrastructures, while still maintaining core H.323 functionality within their networks. With the ability to support the connection of customers and carriers using either rotocol, service providers can offer a variety of application hosting and sharing services, and be more aggressive in pursuing wholesale opportunities via new services. Some principles for coexistence that are critical for successful multiprotocol deployments are transport capabilities across time-division multiplexing (TDM) interfaces, dual tone multifrequency (DTMF) processing capabilities and fax relay support. In deployments where both protocols are used, it is important that there are no performance limitations related to the call mix between SIP and H.323 calls, and that there is no significant deviation in calls-per-second measurements compared to a homogeneous SIP or H.323 network.

Cisco gateways provide support for coexistence of SIP and H.323 calls beginning with Cisco IOS Software Release 12.2(2)XB. Above illustrates packet voice architectures for wholesale call transport and 2 illustrates termination services for application service providers (ASPs) where SIP and H.323 are used simultaneously for signaling.

Reasons for VoIP Deployment

When you are using PSTN line, you typically pay for time used to a PSTN line manager company: more time you stay at phone and more you’ll pay. In addition you couldn’t talk with other that one person at a time.

In opposite with VoIP mechanism you can talk all the time with every person you want (the needed is that other person is also connected to Internet at the same time), as far as you want (money independent) and, in addition, you can talk with many people at the same time.

If you’re still not persuaded you can consider that, at the same time, you can exchange data with people are you talking with, sending images, graphs and videos.

There are two main reasons to use VoIP: lower cost than traditional landline telephone and diverse value-added services.

Low Cost & Higher multimedia application:

Traditional telephone system requires highly trained technicians to install and custom configuration. Companies find the need to call the service of specialist to implement, simple tasks like moving adding a phone. Modules such as ‘voicemail’ and the additional lines are the part of perpetual cycle of upgrades and modifications that make telephony support a very profitable business. The methodology use to implement PSTN business phone system is well understood and the industry is very mature. Hence company can make a purchase with the confidence that if they are installing a traditional system it will function and include an excellent supported infrastructure.

IDC reports the number of VoIP ports shipped in 2005 will be equal to traditional analogues deployment. Non to be taken lightly, the average lifespan of a voice system range from 5-10 years. In 5 to 10 years, an analogues telephone system will be the exception as opposed to the telephone standards. Qualified technicians, whom are required to work on propriety system, will be difficult to come by. In addition, the prospect of telephone manufacture going out of business or the technology simply being repulsed by a more agile and less costly alternative, are both risks that must be taken into account in well informed decision.

Fortunately a company can take few preventive to protect them from outdated system. One such step is use of standards technologies that are back by a number of company and possibly trade group as opposed to a single entity. In VoIP space a good example is session Initiation Protocols, SIP. SIP is supported by the large majority of vendors and is considered the industry standard protocol for VoIP. Beyond analogue lines that terminate from an ISP, The traditional telephony market does not have much interoperability. For example it is not be integrate an Avaya PBX with a Nortel PBX.

Hidden cost can be substantial in any technology deployment. The downtime experienced with buggy or poorly implemented technology, in addition to the cost of qualified consultants to remedy such as

Challenges of VoIP:

Though VoIP is becoming more and more popular, there are still some challenging problems with VoIP:

Bandwidth: Network which available is an important anxiety in network. A network can be busted down into many nodes, associations and produce a big quantity of traffic flow, therefore, the availability of each node and link where we only focus on the bandwidth of the VoIP system. An in a data network, bandwidth overcrowding can cause QoS problems, when network overcrowding occur, packets need to be queued which cause latency as well as jitter. Thus, bandwidth must be accurately reserved and billed to ensure VoIP quality. Because data and voice share the same network bandwidth in a VOIP system, the necessary bandwidth condition and allocation become more complex. In a LAN surroundings, switches usually running at 100 Mbps (or 1000 Mbps), upgrading routers and switches can be the effective ways to address the bandwidth bottleneck inside the LAN.

Power Failure and Backup Systems: Traditional telephones work on 48 volts which is supplied by the telephone line itself without outside power supply. Thus, traditional telephones can still continue to work even when a power breakdown occurs. However, a backup power system is also required with VOIP so that they can continue to operate during a power breakdown. An organization usually has an uninterruptible power system (UPS) for its network to overcome power failure, [14]

Security: As VoIP becomes too popular, the issues related to VoIP network are also very progressively and more arising [15]. W. Chou [16] has investigation the different security of VoIP investigation the different and also given some optional strategies for these issues. In reference [17], the authors also outline the challenges of securing VoIP, and provide guidelines for adopting VoIP technology.

Soft phone: Soft phones are installed on system thus should not be used where the security is an anxiety. In today’s world, worms, viruses, Trojan houses, spy wares and etc are everywhere on the internet and very complex to defend. A computer could be attacked even if a user does not open the email attachment, or a user does nothing but only visit a compromise web site. Thus use of soft phones could bring high risks for vulnerabilities.

Emergency calls: Each traditional telephone link is joined to a physical location, thus emergency tune-up providers can easily track caller’s locality to the emergency send out office. But dissimilar traditional telephone lines, VoIP technology allows an exacting number could be from anywhere; this made emergency services more problematical, because these emergency call centers cannot get the caller’s location or it may not be possible to send out emergency services to that location. Although the VoIP providers provide some solutions for emergency calls, there is at rest need of manufacturing principles in VOIP surroundings.

Physical security: The most significant issue in VoIP network is Physical security. An attacker can do traffic psychoanalysis once actually they access to VoIP. In between server and gateway, like to determine which parties are communicating. So the physical security policy and some controls are needed to control the VoIP network access mechanism. Otherwise, risks such as insertion of snuffer software by attackers could cause data and all voice connections being intercept.

Wireless Security: Connection in wireless network nodes were integrated with VoIP network which receives more and more popular and accepted [18]. The wireless networks are very feeble as compared to Wired Equivalent Privacy (WEP). The algorithm for 802.11 is week because WEP can be cracked with public available software. This is the major project in wireless network for example the more common and popular WiFi protected Access (WPF and WPA 20) which administrated by Wi-Fi Alliance are providing more significant security in improvement, the WPA protected is also integrated with wireless technology in VoIP.

CHAPTER -3

Related Work

3.0 Security Studies

Voice of Internet Protocol is the next generation telecommunications method. It allows to phone calls to be route over a data network thus saving money and offering increased features and productivity. All these benefits come at a price, vulnerability. It is easier to attack and exploit a voice and data network. VoIP will need extra security measures beyond the standard security that is typically implement for a computer network. Many issues need to be addressed such as type of attacks, security, quality of service and VoIP protocols.

Voice over IP (VoIP) is a one of the most challenging technology in today’s market. The importance of VoIP is rapidly growing, many vendors introducing VoIP services with advanced technologies for improving quality of services and security. In this chapter I am discussing about security models and security process.

3.1 VoIP Security Process:

There are many VoIP protocols in the market. Some are proprietary while others are open standards. The two most popular open protocols are H.323 and SIP. They were designed by two different organizations and operate slightly differently. They both have problems with the use of random ports problems with NAT translations and firewalls.

Security for VoIP devices and VoIP network is a complex process, securing of VoIP protocols and data streaming invokes at many stages. The most common VoIP vulnerabilities are as follows

Ø Software Related:

Ø Device related

Ø Protocol related

Ø System Configuration related

Ø Application level attacks

3.1.2 Software Related Vulnerabilities:

The basic flaws in software vulnerable are operating services and functions problems and quality, operating system interface and administrations [19]. Software application interfaces, software application logic

Ø Software applications

Ø Application interfaces

3.1.3 Device Related Vulnerabilities:

One of the most common security threats effects on VoIP hardware devices. In early days the most of the VoIP systems are designed with limited energy power, computing power. Due to the heavy competition in the market many vendors are keeping low cost, they are designing with low cast VoIP hardware devices but due to the changes of software applications, other system infrastructure the system need to regularly updates the device.

The most common hardware devices in VoIP are

Ø PC’s

Ø Telephone adaptors

Ø Modems

Ø VoIP phones

3.1.4 Protocol Vulnerability:

The main protocols in VoIP are H.323 [12] and SIP (Session initiation protocol), these two protocols are commonly used in VoIP hardware system [19]. These protocols overwhelmed with security issues. SIP protocol is a complex protocol which maintains the security in SIP RFC. In SIP the network address translation crack security and which doesn’t examine firewalls.

H.323 is an International Telecommunication Union standard for audio and video communication across a packet network (National Institute of Standards and Technology 2005). There are four types of devices under H.324: terminals, Gateways, Gatekeepers and Multi-Point Conference Units. The terminals are phones and computers. Gateway provides an exit to other networks. The Gatekeeper handles addressing and call routing while the MCU provided conference call support. H.323 uses other protocols to perform other vital tasks. UDP packets using the Real-Time Transport Protocol transport all data. H.225 handles registration, admissions & status, and call signaling. H.235 also handles all security and has four different schemes call Annexes. “H.323 is a complicated protocol”.

SIP Vulnerabilities Overview

The below shows a SIP call flow using SIP and UDP protocols, user can send a voice call through proxy server, the p


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