This paper presents analysis VoIP intrusion detection as well as the techniques suitable for the real-time interactive VoIP communication systems. The paper is basically a proposal that is based on previous studies regarding the same topic. In the paper, it is well provided that effective communication over VoIP is a necessity in the contemporary economy because of the technology’s advantages and benefits over the traditional telephony communication systems. It is also pointed out that its popularity and the advanced features of the VoIP technology, as well as the associated benefits, encourage security threats to companies using VoIP communication systems. Hackers are thus highly attracted by the new technology in an attempt to take advantage of its growth and popularity for personal benefits.
and its features may seemingly attack the attention of hackers as depicted in the paper. In the case of any issues related to data loss or any other related issues, sender-based loss-recovery techniques are used. Nevertheless, the analysis shows that not each of these techniques is suitable for real-time interactive VoIP communications. Previous studies are consulted to strengthen the analysis results. The intervening technique is found to has no impact in achieving consistent improvement within an environment of lost packets. On the contrary other systems such as peer-to-peer VoIP systems are found to have a significant potential for a widespread deployment. The research opens way into studying how to detect Spam in VoIP systems. To enhance successful future implementation of the program, the OPNET 14 Software will be used to construct various simulation scenarios.
VoIP quality of service issues. 8
VoIP is basically an acronym for Voice over IP. It is a methodological approach as well as technologies for delivering multimedia sessions and voice communications over Internet Protocol (IP) networks like the Internet. It can as well be considered as a group of products, which enable advanced communication services usually over data networks (Keromytis, 2012).
VoIP refers to a class of products that enable advanced
communication services over data networks. While voice is
The technology is today emerging as a substitute for the traditional telephone systems and the associated technologies. The invention of this new technology and its benefits has greatly contributed to its growth. The same factors are now contributing to increased hacking problems within the industry. This problem had not however hindered its ever-increasing popularity and deployment. As this goes on, users suffer various problems associated with VoIP threats from various protocol layers.
Session Initiation Protocol is a core protocol for coming real-time communication
networks, including VoIP, IMS and IPTV networks. Based on the open IP stack, it is similarly
susceptible to Denial-of-Service Attacks launched against SIP servers. More than 20
different research works have been published to address SIP-related DoS problems. In this
survey we explain three different types of DoS attacks on SIP networks, called SIP message
payload tampering, SIP message flow tampering and SIP message flooding. We survey
different approaches to counter these three types of attacks. We show that there are
possible solutions for both payload and flow tampering attacks, and partial solutions for
message flooding attacks. We conclude by giving hints how to open flooding attacks issues
could be addressed.
Malicious attackers may exploit the misconfiguration of devices, the vulnerability of the underlying operating systems, and protocol implementation flaws to break in. Well-known attacks of data networks such as worms, viruses, Trojan horse, denial-of-service (DoS) attacks can also plague VoIP network devices. It is claimed that 91% of advanced attack evade and affect network systems . The DoS attacks mainly include Signaling DoS attacks, media DoS attacks, and physical DoS attacks .
An Intrusion Detection System (IDS) helps administrators to monitor and defend against security breaches. Intrusion detection techniques are generally divided into two paradigms, anomaly detection and misuse detection Voice over Internet Protocol (also called VoIP, IP telephony, Internet telephony and Digital Phone) is the routing of voice conversation 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. Typically, the Intrusion Prevention System may be used as an extension of the IDS for an effective protection model by helping prevent malicious attacks . The components of protection model include a Sip server (Asterisk 10.0.0), IDS (Snort 2.9.1), traffic generator (SIPp 3.1), OS (Debian 6.0.2 SIPp), and a free Open Source test tool or traffic generator . It also include free open source network intrusion detection and prevention system. Figure 1 shows the structure of the protective model.
Figure 1: Protection Model
The VoIP is attributed to a number of advantages. Some of the advantages of VoIP include toll bypass, network consolidation and service convergence. Thousands of dollars are saved for large enterprises by placing long-distance calls over an IP network instead of the traditional telephone system. Network consolidation enables the transmission of data, voice, and video over one single network. The integration greatly reduces setup and maintenance costs. With service convergence, enhanced functionality can be implemented through coupling of multimedia services .
The deployment rate of VoIP is increasing steadily. According to Juniper Research, the rapid growth of the global VoIP market will contribute $18 billion in revenues by 2010 . Securing VoIP system is more challenging than securing pure data network. First, all security problems related with data network appear in VoIP system since they share same network infrastructure. Secondly, VoIP does not have a dominant standard so far. The support of two standards in products just increases the chance of buggy application. Dozens of proprietary protocols make the matter worse. Thirdly, the QoS (Quality of Service) requirement of VoIP leaves less working room for possible security measures. A very secure VoIP system that cannot deliver good voice quality is not attractive.
Although VoIP is already widely implemented, the technology is far from mature. There is not a dominant protocol standard in the markets. Every vendor uses either its own proprietary or one of two standards, H.323 and SIP (Session Initiation Protocol). For example, Cisco uses the SCCP (Signaling Connection Control Part) protocol, Avaya uses the H.323 protocol with proprietary extensions and Nortel uses the UNISTIM (Unified Network Stimulus) protocol. Proprietary protocols make it difficult to interconnect products from different vendors. Good news is that more and more vendors begin to support H.323 or SIP. VoIP transfer voice signal over IP based network. First, human voice needs to be converted to digital bits and encapsulated in packets, which are transmitted using IP network and converted back to voice signal at the destination. Secondly, there should be a way to identify each entity in the network like phone number in the traditional telephone system. Thirdly, VoIP entities need to be able to communicate with telephones of PSTN. Generally, implementation, the OPNET 14 Software could be necessary for successful implementation in order to reflect a way of solving the reflected problems and achieve possible advantages of VoIP against the traditional PSTN .
Generally, there are four major problems associated with VoIP instruction detection. These problems include confidentiality threats, eavesdropping of phone conversation, unauthorized access attacks, and cell redirection or hijacking. Each of these problems is described below.
Confidentiality implies that any given information cannot be accessed by unauthorized parties. The confidential information of end-users includes private documentation, financial information, and security information like password, conversion content, conversion history or pattern among others. The confidential information for network components includes operation systems, IP addresses, protocols used address mapping, and user records among others. Leak of this information might make attackers’ jobs easier.
Eavesdropping of phone conversation
Conventional telephone eavesdropping requires either physical access to tap a line, or penetration of a switch. With VoIP, opportunities for eavesdroppers increase dramatically because of the large number of nodes in the path between two conservation entities. If the attacker compromises any of these nodes, he can access the IP packets flowing through that node. There are many free network analyzers and packet capture tools that can convert VoIP traffic to wave files . These tools allow the attackers to save the conversation into the files and play them back on a computer. VoMIT or Voice over Misconfigured Internet Telephones is an example of such a tool . Ethreal can also be used to record SIP packets and retrieve voice message in wav file format.
Unauthorized access attack
Unauthorized access implies that the attacker(s) can access resources on a network that they do not have the authority. Shawn Merdinger reported multiple undocumented ports and services in certain VoIP phones . There are also vulnerabilities due to implementation issues such the lack of familiarity .
There are systems for call control, administration, billing and other voice telephone functions. Repositories in these systems may contain passwords, user identities, phone numbers, and private personal information. Lots of gate ways and switches are shipped with default well-known passwords. If these passwords are left without changes, the attacker scan easily break in. Some switches still use TELNET for remote access. The clear-text protocol exposes everything to anyone who can sniff the network traffic. Some of the gate ways or switches might have a web server interfaces for remote control. The attacker might sniff the HTTP traffic in local network to steal sensitive information. Attacker scan also use ARP cache poisoning to forward all traffic through their machines to capture network traffic .
Encryption of voice message packets can protect against eavesdropping. IPSec can be deployed to encrypt whole packets. SRTP can provide confidentiality, message authentication and replay protection for audio and video streams. To better protect gateways and switches, they should use SSH instead of other clear-text protocols as remote access protocol. If web-based interface is provided, HTTPS should replace HTTP. In addition, all default passwords should be changed before the system is plugged into the network. An up-to-date intrusion detection system might detect ARP poisoning and other types of attacks .
Integrity of information means that information remains unaltered by unauthorized users. A legitimate user may perform an incorrect or unauthorized operations function and may cause delirious modification, destruction, deletion or disclosure of switch software and data. An intruder may masquerade as a legitimate user and access an operation port of the switch. Registration hijacking happens when an attacker replaces the legitimate registration of the victim with his address. The attack causes all incoming calls for the victim to be sent to the attacker’s address. Proxy impersonation attack tricks the victim into communicating with a rogue proxy set up by the attacker. Once an attacker impersonates a proxy, he has complete control of the call. Figure 8 illustrates proxy impersonation. The attacker tricks Alice to communicate with the rogue proxy server instead of the legitimate proxy server . The UAs and proxies normally communicate using UDP and do not require strong authentication to communicate with another proxy. The attack can work by several means, including DNS (Domain Name Service) spoofing, ARP (Address Resolution Protocol) cache spoofing, DHCP spoofing, or changing proxy address for a SIP phone.
Call redirection or hijacking
Call redirection happens whenever calls are intercepted and rerouted through different paths before getting to the destination. Potential methods in this regard include proxy impersonation as well as registration spoofing. The attackers could spoof the response mainly from the call recipient and then trick the call requestor into talking with the attacker. Three classes of protection goals basically exist although also other classifications are possible. The first, which is also the most intuitive class in this case, is confidentiality. This refers to the protection against unwanted disclosure of information. Confidentiality could be additionally subdivided into data confidentiality, which implies to keeping the secrecy of payload data; confidentiality of the identity of the data’s owner, which refers to anonymity; confidentiality of the pure communication fact; and confidentiality of network structure. The goals here are meant to eliminate or at least minimize the issues of confidentiality, availability, and integrity . Anonymity of a given entity against its communication partner or the communication partners against any third parties could be required.
A more specific subclass is the confidentiality of the pure fact of communication. It means that an outside observer is not aware that communication is taking place at all. The fourth subclass, confidentiality of the network structure, is due to the fact that network providers want to conceal properties of their own network such as size, topology, and number of connected customers/nodes, host names, network types, network capacity, availability from competitors while still being able to exchange data and allow roaming. The second major class is integrity of data . Assuring integrity of data means guaranteeing that data is not deleted or replaced along the communication path. Prevention of unauthorized modification of data is called strong integrity protection whereas weak integrity protection only ensures that modifications can be detected. Integrity may also refer to the identities of a communication partner .
Assuring integrity of identity means making it impossible for an attacker to use a fake address and masquerade as someone else. Otherwise, it would be possible that confidential data is sent to the attacker instead of the intended recipient. This in turn leads to a violation of the protection goal of data confidentiality. Using a fake identity, also known as spoofing, is difficult in PSTN but common in IP based networks. The third class is security requirements concerning availability. The attacks against availability are known as Denial of Service (DoS). Such an attack in a PSTN may be the physical disconnection of a network or power cable, gaining administrator access and shutting down the system or deleting files, etc. In case of IP based networks, the above mentioned and also numerous other DoS attacks are possible. An example would be degrading a service such as VoIP by sending a great number of forged requests to key network elements, e.g. the call processing server. Generally, some integrated security solutions in SIP can be employed to solve some noticeable threats . Table 1 shows possible threats and their corresponding integrated security solutions in SIP.
Table 1: Integrated Security Solutions in SIP
|Call hijacking/man in the middle||Authentication mechanisms|
|Tampering the audio or the signaling||Integrity and non repudiation
2.0 LITERATURE REVIEW
In the curse of designing security mechanisms one has to determine the degree at which every security requirements has to be fulfilled. This aspect depends on the attack types and the profile of the attacker. Different types of attackers are known to have different motives for performing attacks. The attacker could be a student with a goal of having fun snooping on people’s e-mails, crackers test out people’s security systems, ex-employees getting revenge for having been fired, or even spies learning enemies military or industrial secrets among others .
Regarding types of attacks, they may be classified in two broad ways . The first way being to distinguish between passive attacks and active attacks whereby a “passive attack” would refer to an attempt to learn or make use of information obtained from the system, although it does not affect any system resources. On the other hand an “active attack”, tries to modify system resources or even affect their operation. Active attacks mainly include interruption of services, modification of messages, fabrication of messages, and replay to attacks. The latter implies to the recording of legitimate messages and sending them again and again mainly to the recipients . In order to achieve a predetermined protection goal different protection mechanisms are applied. Confidentiality of data is achieved by installing network components specifically in physically secure environments or even by using encryptions. Encryption may be used for providing data confidentiality and for protecting data integrity through means of hash values, Message Authentication Code (MAC) or digital signatures.
2.2 Types of Attacks in Communication Networks
An important issue for achieving most protection goals is providing access control, i.e. allowing only authorized users to gain access to the system. Therefore, first the identity of the user requesting access has to be verified which is called authentication. Then the request, e.g. placing a telephone call, is used together with the user ID and a policy database to check whether the user should be granted or rejected access to the system. This is called authorization. A related issue is accounting, i.e. requiring payment from the user according to the service used. Authentication, authorization and accounting mechanisms, known as AAA, are an essential part of every communication network, including ISDN, GSM, UMTS and IP based networks .
Implementing the security mechanisms introduced in this case can be done on each layer of the network protocol stack: physical, data link, network, and transport or application layer. Another very important aspect is that even providing a good technical solution is still not sufficient. Giving the opportunity for malicious persons to operate the system on his behalf? In fact, most damage due to successful attacks against security is the result of human mistakes rather than technical weaknesses of the systems .
It is typically crucial to make users aware of security issues to minimize this source of security threats. This is called “social engineering”. It may not be discussed in much detail as this article is focused primarily on the technical side of security but understanding its importance is essential. Therefore, more countermeasures against these and other attacks are necessary. However, stating an important note is appropriate at this place . In practice, no perfect solution for securing a system may be found which is able to prevent all existing (and not yet existing) attacks.
Aiming an absolute perfection will inevitably lead to an increase of costs and complexity and will cause much inconvenience for the users interacting with the system. Therefore it is important to find a trade-off between security services, the efforts to provide these services (e.g. equipment, implementation and maintenance costs) and the level of inconvenience that users will experience while interacting with the system due to the additional security services. In order to find this trade-off the designer should first clarify against which attackers the system should offer protection and what means they have for attacking the system. Particularly for the design of an IP Telephony network the following aspects should be considered when searching for an optimal trade-off .
A segmentation of the data and voice is a key issue. This may seem to be in conflict with the main idea of VoIP which is the integration of voice into an existing IP data network. However, segmentation does not mean that two different IP infrastructures should be deployed: one for voice and one for data. A logical separation in voice and data segments can be done instead by the use of VLANs. For creating a VLAN the layer 2 switches have to be VLAN aware; that is, they should have the required functionality to support VLANs. The network administrator can then assign each device on the network to a certain VLAN. Thereby it should be considered that many IP phones support a data port to allow the connection of a PC to the phone so that only a single cable is necessary to provide data and voice connectivity to the user. Hence, these should also have options to support VLAN. .
2.3 VoIP Techniques
Researchers have proposed many techniques for improving quality of service (QoS) in the face of packet loss. Some of these techniques employ receiver-based packet-loss concealment (PLC) approaches. Many audio encoder/decoders use PLC algorithms to synthesize audio when packets of audio data are missing. There are also sender-based loss-recovery techniques, whereby the sender assumes an active role to help the receiver recover lost data or improve Qu’s when packet loss occurs. Commonly, sender-based techniques are independent of receiver-based techniques, so designers can employ both types of loss-recovery methods simultaneously .
The choice of any particular audio encoder/decoder (i.e., codec) with PLC transparently provides the degree of receiver-based packet-loss recovery that the selected codec offers. Designers normally consider packet-loss resilience along with several other factors when choosing codecs. Sender-based packet-loss recovery can supplement or even entirely replace receiver-based packet-loss concealment, so designers must also consider sender-based packet-loss recovery as part of the overall design for improving QoS. Most sender-based loss-recovery mechanisms work by retransmitting data or by transmitting additional data .
These approaches consume additional resources such as network bandwidth and CPU capacity. The increase in the consumption of network bandwidth puts more loads on the network and can potentially result in the loss of more packets, perhaps ironically exacerbating the very problem we are trying to solve. Sender-based loss-recovery techniques typically introduce added end-to-end delay into the media stream. Generally, humans cannot even perceive a one-way delay of less than 100ms, and most users can tolerate a one-way delay of up to 250ms.If the one-way delay exceeds 250 ms, however, the delay can result in a serious talker-overlap effect that is intolerable for most users .
2.3.1 Loss-Recovery Techniques
When trying to combat the effects of packet loss in VoIP systems, designers typically do not use sender-based loss-recovery techniques. One of the main reasons for avoiding sender based loss-recovery techniques is that all of the existing approaches require additional network bandwidth. As the utilization of the data bandwidth increases, the data network becomes more congested and can discard more packets. Consequently, sender-based loss-recovery techniques, which are supposed to improve QoS when packet loss occurs, can actually degrade the audio quality by causing more lost packets .
Plain delivery is more prevalent than any other delivery technique in VoIP solutions. Plain delivery does not provide any sender-based effort to improve audio quality when packet loss occurs. We include this approach to provide a baseline for purposes of comparison. This technique simply packages each block of encoded audio data into an IP packet and transmits the packet. For example, if we use a 20 mms frame length; we package 38 bytes of iLBC encoded data into an IP packet for transmission. In the case of G.729A, we package 20 bytes of encoded audio data for a 20 ms period into an IP packet for transmission .
In either case, we transmit 50 packets per second. If we transmit the packets via Ethernet, the header overhead is 78 bytes per IP packet (12 bytes for the interpacket idle time, 26 bytes for Ethernet overhead including CRC and preamble, 20 bytes for IPv4, 8 bytes for UDP, and 12 bytes for RTP). Consequently, we consume 46.4 kb/s for iLBC and 39.2kb/s for G.729A. Plain delivery introduces 20 ms of delay into the audio stream. Interleaving Interleaving  is not really a loss-recovery technique, since this approach is not able to recover any lost data despite its importance . However, this method attempts to reduce the degradation of perceptual audio quality by distributing lost data into several small gaps instead of having one large gap of lost data.
Many researchers believe that listeners can mentally patch over a loss more easily if we disperse the loss into several small parts. Since this technique does not transmit additional information, it requires the same bandwidth utilization that plain delivery uses. Interleaving is feasible only if we transmit multiple frames of audio in each IP packet . If we packaged two frames of audio into one IP packet, we would transmit packets of interleaved audio frames, as Fig. 1 illustrates. We could further scatter lost frames by interleaving more frames into each IP packet. Figure 2 illustrates interleaving with four audio frames per IP packets.
Increased interleaving provides improved dispersal of lost audio frames, but increased interleaving also introduces a larger delay into the media stream. Larger delays can cause serious problems such as talker overlap, so interleaving improves some aspects of quality while degrading others. Forward error correction (FEC) is a sender-based technique for mitigating the undesired effects of packet loss. FEC works by transmitting redundant packets for error correction. Reed–Solomon encoding is a well-known block-based error-correction mechanism. Reed-Solomon codes enjoy widespread use in storage applications and digital communications ranging from Compact Disk and DVD to wireless communication, satellite communication, digital television, and high-speed modems .
The Reed–Solomon encoding scheme, works by generating parity bits and the sending the parity bits along with the data values. If data values are missing or corrupt, the Reed–Solomon decoder cans reconstruct the original data by using the redundant information from the parity bits. The measure of the redundancy in the block determines the error-correcting ability of the Reed–Solomon code. Reed–Solomon coding can be computationally quite expensive to implement on a general-purpose microprocessor, especially in a real-time software implementation such as VoIP. A router drops packets when it is overloaded, so a router is likely to drop packets that arrive during a busy time. A router that drops the original packet is therefore likely to drop the duplicate packet as well if we transmit the duplicate packet immediately after the original packet . If we delay the transmission of the duplicate packet, we reduce the chance of losing both packets but also need to add this delay into the overall delay that this approach introduces into the audio stream.
2.4 VoIP Quality of Service Issues
Jitter refers to non-uniform packet delays. It is often caused by low bandwidth situations in VOIP and can be exceptionally detrimental to the overall QoS. Variations in delays can be more detrimental to QoS than the actual delays themselves. Jitter can cause packets to arrive and be processed out of sequence.
Latency in VOIP refers to the time it takes for a voice transmission to go from its source to its destination. Ideally, we would like to keep latency as low as possible but there are practical lower bounds on the delay of VOIP. Packet LosPacket loss is another major QoS issue for VoIP systems. VOIP is exceptionally intolerant of packet loss. Packet loss can result from excess latency, where a group of packets arrives late and must be discarded in favor of newer ones. It can also be the result of jitter, that is, when a packet arrives after its surrounding packets have been flushed from the buffer, making the received packet useless.
Administrators must be able to prioritize allocated bandwidth to VoIP traffic to ensure adequate quality of service (QoS). Underperforming firewalls can potentially degrade VoIP communications with latency, jitter, packet loss and echo. The ideal solution would include control over QoS prioritization, comprehensive security that introduces no latency, and secure access to VoIP applications over VPNs. QoS network security appliances provide built-in, security and bandwidth optimization, as well as easy support for VoIP, SIP devices throughout the network. Integrated QoS allows prioritization of VoIP and data traffic on the network.
Some of the IDSs are generic in nature and can be customized with detection rules specific to the environment in which they are deployed (e.g., Snort and Prelude Most existing IDS solutions, such as Snort, provide only a generic architecture in which the IDS itself does not pay much attention to the application level protocols and the corresponding state information. Consequently, the user has to construct detection rules from the ground to capture these application level contexts for performing detection. A traditional IDS like Snort with a rule to detect multiple 4XX responses may flag a large number of false alarms. The Snort IDS is well known for its efficiency in examining incoming packets and SPACEDIVE leverages the Snort functionality.
To avoid performance loss, SPACEDIVE is built into Snort using part of its low-level functionality (examining and processing packets) and adding to it (e.g., to build state to support stately detection) and building completely the high level functionality specific to the VoIP environment. The native rule language of Snort is not well-suited for VoIP stateful or cross-protocol detection. Snort provides limited capability for remembering state both within a VoIP session for a given protocol (e.g. SIP) and across protocols (e.g. SIP & RTP). To make up for this, we add constructs to the existing rule language so that it is better-suited for detecting attacks targeted to VoIP environments that span packets in a session and different protocols.
Our analysis shows that not every sender-based loss-recovery technique is suitable for real-time interactive VoIP communications. Although previous researchers have reported that interleaving can reduce the degradation of perceptual audio quality, our study shows that the interleaving technique does not achieve any consistent or significant improvement in an environment of lost packets. Another important area of our ongoing research is that of peer-to-peer VoIP systems. Peer-to-peer VoIP systems have a great potential for wide-spread deployment – an example is the popular Skype system. We are also looking at ways to detect Spam in VoIP systems. Finally, we are investigating the effectiveness of our system in presence of secure protocols. For successful future implementation, the OPNET 14 Software will be used to construct various simulation scenarios. The simulation is to ensure successful implementation in order to reflect the possible advantages of VoIP against the traditional PSTN.
Concerning the types of attacks these may be classified in two ways. The first way is to distinguish between passive and active attacks. A “passive attack” is an attempt to learn or make use of information from the system but does not affect system resources an “active attack”, on the other hand, attempts to alter system resources or affect their operation. Active attacks include the interruption of a service, the modification of messages, the fabrication of messages (often with fake source addresses) and replay attacks, i.e. recording a legitimate message and sending it again and again to the recipient.
Based on the analysis and consulted literature, it can be pointed out that achievement of protection goals would always required the use of different protection mechanisms. Confidentiality of the data is mostly achievable through installing network components specifically in physically secure environments or it could be achieved by using encryption as explained in the report. The encryption could be applied for providing confidentiality of data as well as in protecting data integrity by means of hash values, Message Authentication Code (MAC) or digital signatures.
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We have highlighted some of the most popular subjects we handle above. Those are just a tip of the iceberg. We deal in all academic disciplines since our writers are as diverse. They have been drawn from across all disciplines, and orders are assigned to those writers believed to be the best in the field. In a nutshell, there is no task we cannot handle; all you need to do is place your order with us. As long as your instructions are clear, just trust we shall deliver irrespective of the discipline.
Are your writers competent enough to handle my paper?
Our essay writers are graduates with bachelor's, masters, Ph.D., and doctorate degrees in various subjects. The minimum requirement to be an essay writer with our essay writing service is to have a college degree. All our academic writers have a minimum of two years of academic writing. We have a stringent recruitment process to ensure that we get only the most competent essay writers in the industry. We also ensure that the writers are handsomely compensated for their value. The majority of our writers are native English speakers. As such, the fluency of language and grammar is impeccable.
What if I don’t like the paper?
There is a very low likelihood that you won’t like the paper.
- When assigning your order, we match the paper’s discipline with the writer’s field/specialization. Since all our writers are graduates, we match the paper’s subject with the field the writer studied. For instance, if it’s a nursing paper, only a nursing graduate and writer will handle it. Furthermore, all our writers have academic writing experience and top-notch research skills.
- We have a quality assurance that reviews the paper before it gets to you. As such, we ensure that you get a paper that meets the required standard and will most definitely make the grade.
In the event that you don’t like your paper:
- The writer will revise the paper up to your pleasing. You have unlimited revisions. You simply need to highlight what specifically you don’t like about the paper, and the writer will make the amendments. The paper will be revised until you are satisfied. Revisions are free of charge
- We will have a different writer write the paper from scratch.
- Last resort, if the above does not work, we will refund your money.
Will the professor find out I didn’t write the paper myself?
Not at all. All papers are written from scratch. There is no way your tutor or instructor will realize that you did not write the paper yourself. In fact, we recommend using our assignment help services for consistent results.
What if the paper is plagiarized?
We check all papers for plagiarism before we submit them. We use powerful plagiarism checking software such as SafeAssign, LopesWrite, and Turnitin. We also upload the plagiarism report so that you can review it. We understand that plagiarism is academic suicide. We would not take the risk of submitting plagiarized work and jeopardize your academic journey. Furthermore, we do not sell or use prewritten papers, and each paper is written from scratch.
When will I get my paper?
You determine when you get the paper by setting the deadline when placing the order. All papers are delivered within the deadline. We are well aware that we operate in a time-sensitive industry. As such, we have laid out strategies to ensure that the client receives the paper on time and they never miss the deadline. We understand that papers that are submitted late have some points deducted. We do not want you to miss any points due to late submission. We work on beating deadlines by huge margins in order to ensure that you have ample time to review the paper before you submit it.
Will anyone find out that I used your services?
We have a privacy and confidentiality policy that guides our work. We NEVER share any customer information with third parties. Noone will ever know that you used our assignment help services. It’s only between you and us. We are bound by our policies to protect the customer’s identity and information. All your information, such as your names, phone number, email, order information, and so on, are protected. We have robust security systems that ensure that your data is protected. Hacking our systems is close to impossible, and it has never happened.
How our Assignment Help Service Works
1. Place an order
You fill all the paper instructions in the order form. Make sure you include all the helpful materials so that our academic writers can deliver the perfect paper. It will also help to eliminate unnecessary revisions.
2. Pay for the order
Proceed to pay for the paper so that it can be assigned to one of our expert academic writers. The paper subject is matched with the writer’s area of specialization.
3. Track the progress
You communicate with the writer and know about the progress of the paper. The client can ask the writer for drafts of the paper. The client can upload extra material and include additional instructions from the lecturer. Receive a paper.
4. Download the paper
The paper is sent to your email and uploaded to your personal account. You also get a plagiarism report attached to your paper.
PLACE THIS ORDER OR A SIMILAR ORDER WITH US TODAY AND GET A PERFECT SCORE!!!