Voice Over Internet Protocol Network Research Articles

Dynamic voice spammers detection using Hidden Markov Model for Voice over Internet Protocol network

Voice over Internet Protocol (VoIP) provides flexible and cost effective services. These services are used by voice spammers to generate unsolicited voice calls. Earlier research declared that the VoIP services are misused for making prank calls, product promotion calls and credit card services. This results in customer dissatisfaction and the financial losses in the bank and telecommunication sectors. Thus, detection of spammers is an essential task to enhance the quality of services in VoIP network. Spammers occur in the form of either human spammer or computer spammer and mimic as the legitimate caller. Voice spammer's states in successive time period are dynamic and dependent, particularly the human voice spammer exhibit high degree of dynamism. This poses challenge for traditional spam detection algorithms. In this paper, a Dynamic Voice Spammer Detection Model (DVSDM) based on the Hidden Markov Model (HMM) is proposed. This model estimates voice spammer's states by using various behaviour variables and detects the voice spammers before reaching the victim. The performance of this detection model is experimentally evaluated with two scenarios (mild and heavy distribution of voice spam calls). The proposed model achieves a False Positive Rate (FPR) of less than 2% and 5% for heavy and mild distribution of voice spam calls respectively. Moreover, the DVSDM model achieves a True Positive Rate (TPR) of 95% for heavy and 92% for mild distribution of voice spam calls.

An optimized hardware calibration technique for transmission of real-time applications in VoIP network

Voice over Internet Protocol (VoIP) uses packet switching to transmit voice, video, data and chat services. VoIP applications depend upon various speech codecs which vary in speech quality, bandwidth and computational requirements. The key challenges for improving the VoIP Quality of Service (QoS) and Quality of Experience (QoE) is to use best codec by considering network traffic and selecting the optimal hardware. Hardware Calibration is a mechanism to aid service providers by selecting a suitable hardware for Call Manager to transmit different real-time based applications. The paper focuses on proposing a computational model which suggests the most suitable hardware for a specific load handling requirement. The selection of processor is done by taking into consideration the effect of different codecs when it takes calls per second as input and gives output as whichever hardware is recommended for Call Manager to handle different types of applications like Peer-To-Peer (P2P) application, Back-To-Back (BTB) application, enabling voice logging, etc. The emulations demonstrate that the proposed model is suitable for selection of hardware which provides better QoS and QoE for the transmission VoIP based real-time applications. Further, a multi-criteria decision-making method based on Analytic Hierarchy Process (AHP) is also proposed to decide which hardware can be chosen as Call Manager. The results obtained from AHP are used to validate the outcome of the proposed computational model. The computational model is applicable only when the tests are run on particular systems and it suggests the best hardware among them for the particular offered load. However, to suggest the best configuration corresponding to the input load from the possible configurations available in the market, benchmarking is performed. A common scale model for hardware benchmark estimation particularly has been proposed for different scenarios.

A scalable detection and prevention scheme for voice over internet protocol (VoIP) signaling attacks using handler with Bloom filter

SummaryIn this paper, a two‐tier model has been developed that includes a Handler and a Bloom filter (HBF). In the first‐tier, the handler detects both the flooding and fake signaling attacks. The Bloom filter, in the second‐tier, prevents both the attacks before reaching the victim. In the existing systems, the packet level features are used which do not perform well for detection and prevention of both the attacks. In this work, flow level features are applied in both tiers. The proposed model is implemented on the innocent Session Initiation Protocol (SIP) server in the VoIP network. The two‐tier model ensures the reliability and trustworthiness between the service provider and the customer. Besides, it also provides billing information along with the exact call duration to a customer who makes a call. The experimental results show that the HBF results in a reduced detection time of 9 seconds with the reduced false positive (FP) of less than 1% and the false negative (FN) of 0.002% and also preserves the voice call quality during media conversation.

Analisa Quality Of Service Dan Implementasi Voice Over Internet Protocol Dengan Menggunakan IPSEC VPN

At present technology, specifically in telecommunication sector, develops very rapidly, and mainly adopts Internet Protocol (IP) to support the communication network. In IP networks it involves voice communication commonly called Voice over Internet Protocol (VoIP). It the development of telephone communication using IP networks as paths of data transfer. As it is an open source, its network security becomes a crucial issues in VoIP. One of the ways to cope with the security problems is to adopt Intenet Protocol Security (IPSec) in VoIP networks. The communication trials used two different bandwidths – 375 kbps and minimum bandwidth standard – codec G.711 u. The result of the analysis showed that the trial using minimum bandwidth standard – codec G.711 u was not suitable to be applied in VoIP IPSec network, as in average it had package loss by 58.75%. Accordingly, the VoIP IPSec networks with 375 kbps bandwidth were more superior. In average, with delay rate (3,4956 ms), jitter (6,8894 ms) and package loss (0,00%). In conclusion, VoIP IPSec networks requires large bsndwidth, i.e. : minimally 375 kbps in its operation.

Kerberos: A real-time fraud detection system for IMS-enabled VoIP networks

In this paper we present the design, implementation and experimental evaluation of Kerberos, an architecture for the detection of frauds in current generation Voice over IP (VoIP) networks. Kerberos is fed by an On-line Charging System (OCS) generating events associated with the setup, evolution and tear-down of end-user calls in a VoIP network compliant with the IP Multimedia Subsystem (IMS) specification. Such events are properly correlated in order to identify, in real-time, patterns associated with a fraudulent utilization of the Operator's resources. The detection phase can in turn trigger the subsequent remediation actions. Communication between the OCS and Kerberos is based on an asynchronous paradigm, whereas event correlation and analysis are effectively realized through a Complex Event Processing approach. The paper will shed light on both the design and the implementation of the system, whose performance is then evaluated by relying on a real-world dataset of Call Detail Record (CDR) events provided by Tiscali, a well known Italian Operator.

A Scalable Detection Technique for Real-time Transport Protocol (RTP) Flooding Attacks in VoIP Network

Voice over Internet Protocol (VoIP) has become widely uses RTP as its media protocol that delivers multimedia sessions over IP based network. RTP is the hub of media flooding attacks that causes inconsistency of media transmitted which degrading Quality of Service (QoS) and unnecessary resource consumption in servers. Thus, the motto of the paper is revolved around detection of RTP flooding attacks. In this paper, Packet Inspection using Statistical (PIS) technique is proposed that detects RTP flooding attacks. The experimental results show that the proposed PIS technique has a detection rate of more than 98% with False Positive (FP) of less than 0.1%.

Voice over quantum IP routing based on least relay node constrained optimization strategy

Quantum communication is the interdisciplinary science of quantum mechanics and telecommunication theory. It has advantages of perfect information security and high efficiency in transmission. In recent years, the theoretical and experimental results show that quantum communication systems have the superiority over the traditional communication systems. Quantum communication systems are hopeful for solving the information security problems that everyone is facing today, therefore, they possess broad application prospects and are forming a research hotspot of the telecommunications field currently. On the other hand, Voice over Internet Protocol (VoIP) is the method to transmit the digitized packet voice in Internet around the world. The advantages of VoIP are that it can carry voice, data, video, telephone conference, electronic commerce, and electronic mail economically. VoIP can realize the information storage and retransmission easily and flexibly. However, VoIP also encounters the problem of information security. We are trying to combine the quantum communications network and the VoIP system together and build a brand new network named quantum VoIP network which combines the advantages of both quantum communications and VoIP. The data packets may be delayed and lost in a queue up with a router due to the congestion and link failure during the transmission of quantum information. In order to ensure the performance of quantum VoIP system, the routing optimization strategies are proposed in the paper. The relay technology based on entanglement swapping is adopted. The multiuse quantum communications are realized by giving priority to the quantum channels with the least relay nodes. Theoretical analysis and simulation results show that when the data transmission links are fail to work properly and routers are in congestion, adopting the routing optimization strategies in M/M/m queuing system with the bit error rate (BER) of quantum bit setting to be 0.2 and the number of common channels increasing from 4 to 8,, the percentage of call failure in quantum communication network decreases from 0.25 to 0.024, and the maximum throughput of quantum networks increases from 64 kbps to 132 kbps. In comparison, when the number of common channels is set to be 4 andthe BER of the quantum bit is from 0.3 to 0.1, the maximum throughput of quantum networks increases from 41 kbps to 140 kbps. Thus it can be concluded that the routing optimization strategies proposed in this paper can improve the performance of quantum VoIP system significantly.

An energy efficient authenticated key agreement protocol for SIP-based green VoIP networks

Voice over Internet Protocol (VoIP) is spreading across the market rapidly due to its characteristics such as low cost, flexibility implementation, versatility of new applications, etc. However, the voice packets transmitted over the Internet are not protected in most VoIP environments, and then the user׳s information could be easily compromised by various malicious attacks. So an energy-efficient authenticated key agreement protocol for Session Initial Protocol (SIP) should be provided to ensure the confidentiality and integrity of data communications over VoIP networks. To simplify the authentication process, several protocols adopt a verification table to achieve mutual authentication, but the protocols require the SIP server to maintain a large verification table which not only increases energy consumption but also leads to some security issues. Although several attempts have been made to address the intractable problems, designing an energy-efficient authenticated key agreement protocol for SIP-based green VoIP networks is still a challenging task. In this study, we propose an efficient authentication protocol for SIP by using smartcards based on elliptic curve cryptography. With the proposed protocol, the SIP server needs not to store a password or verification table in its database, and so no energy is required for the maintenance of the verification table. Security analysis demonstrates that the proposed protocol can resist various attacks and provides efficient password updating. Furthermore, the experimental results show that the proposed protocol increases efficiency in comparison with other related protocols.

SIP Flooding Attacks Detection and Prevention Using Shannon, Renyi and Tsallis Entropy

Voice over IP (VOIP) network, also known as Internet telephony, is growing increasingly having occupied a large part of the communications market. With the growth of each technology, the related security issues become of particular importance. Taking advantage of this technology in different environments with numerous features put at our disposal, there arises an increasing need to address the security threats. Being IP-based and playing a signaling role in VOIP networks, Session Initiation Protocol (SIP) lets the invaders use weaknesses of the protocol to disable VOIP service. One of the most important threats is denial of service attack, a branch of which in this article we have discussed as flooding attacks. These attacks make server resources wasted and deprive it from delivering service to authorized users. Distributed denial of service attacks and attacks with a low rate can mislead many attack detection mechanisms. In this paper, we introduce a mechanism which not only detects distributed denial of service attacks and low rate attacks, but can also identify the attackers accurately. We detect and prevent flooding attacks in SIP protocol using Shannon (FDP-S), Renyi (FDP-R) and Tsallis (FDP-T) entropy. We conducted an experiment to compare the percentage of detection and rate of false alarm messages using any of the Shannon, Renyi and Tsallis entropy as a measure of disorder . Implementation results show that, according to the parametric nature of the Renyi and Tsallis entropy, by changing the parameters, different detection percentages and false alarm rates will be gained with the possibility to adjust the sensitivity of the detection mechanism.

Vtalk: Secure Deployment of VoIP over LAN’S

Peer-to-peer VoIP (voice over IP) networks, exemplified by Skype, are becoming increasingly popular due to their significant cost advantage and richer call forwarding features than traditional public switched telephone networks. One of the most important features of a VoIP network is privacy (for VoIP clients). Unfortunately, most peer-to-peer VoIP networks neither provide personalization nor guarantee a quantifiable privacy level. In this paper, we propose novel flow analysis attacks that demonstrate the vulnerabilities of peer-to-peer VoIP networks to privacy attacks. We then address two important challenges in designing privacy-aware VoIP networks: Can we provide personalized privacy guarantees for VoIP clients that allow them to select privacy requirements on a per-call basis? How to design VoIP protocols to support customizable privacy guarantee? This paper proposes practical solutions to address these challenges using a quantifiable k-anonymity metric and a privacy-aware VoIP route setup and route maintenance protocols. We present detailed experimental evaluation that demonstrates the performance and scalability of our protocol, while meeting customizable privacy guarantees. IndexTerms: VOIP,SSH,QOS,ITU,RSP,AARSP,FCC,PSTN

Stochastic modeling for delay analysis of a VoIP network

VoIP (“Voice Over Internet Protocol”) is the transmission of voice communication through the Internet via IP-based telephony networks. VoIP has become very popular over recent years due to the cost advantages for consumers and businesses compared to the traditional telephony networks. Since it is deployed on packet-based networks, one of the major Quality of Service (QoS) concerns of VoIP technology is the average end-to-end connection delay. The objective of this paper is to present a queuing model for obtaining the end-to-end delay of a VoIP connection. The paper first describes all the partial delay components, and their mathematical formulations. Subsequently, based on all the partial delay components, a queuing model for the end-to-end delay of a VoIP connection is presented. The proposed queueing model is analyzed using a generalized stochastic Petri net (GSPN) model. From the GSPN model, we obtain numerical results for the end-to-end delay, which are presented graphically. The results are in accordance with the expected behavior of delay in a VoIP network.

Caller-REP: Detecting unwanted calls with caller social strength

Voice over IP (VoIP) is a cost effective mechanism for telemarketers and criminals to generate bulk spam calls. A challenge in managing a VoIP network is to detect spam calls without user involvement or content analysis. In this paper we present a novel content independent, non-intrusive approach based on caller trust and reputation to block spam callers in a VoIP network. Our approach uses call duration, interaction rate, and caller out-degree distribution to establish a trust network between VoIP users and computes the global reputation of a caller across the network. Our approach uses historical information for automatically determining a global reputation threshold below which a caller is declared as socially non-connected and as a spammer. No VoIP data-set is available for testing the detection mechanism. We verify the accuracy of our approach with synthetic data that we generate by randomly varying the call duration, call rate, and out-degree distributions of spammers and legitimate users. This evaluation shows that our approach can automatically detect spam callers in a network. Our approach achieves a false positive rate of less than 10% and true positive rate of almost 80% in the first two days even in the presence of a significant number of spammers. This increases to a true positive rate of 99% and drops a false positive rate to less than 2% on the third day. In a network with no spammers, our approach achieves a false positive rate of less than 10%. In a network heavily saturated with more than 60% of spam callers, our approach achieves a true positive rate of 98% and no false positives. We compare the performance of our approach with a closely related spam detection approach named Call-Rank. The results show that our approach outperforms Call-Rank in terms of detection accuracy and detection time.

Design and implementation of a reconfigurable test-bed for real-time security measurements in VoIP systems

Voice over Internet Protocol (VoIP) systems have to face a great number of security threats, which are basically connected to the underlying Internet Protocol (IP) infrastructure. Along with relevant economic losses and malfunction for the VoIP system itself, external attacks exploiting a security hole could also bring to a major security failure for the interconnected Public-Switched Telephone Network (PSTN), usually regarded as a reliable and secure domain. This is the reason why before connecting a private VoIP network to the carrier infrastructure, a great number of tests have to be carried out, to verify the security robustness of the VoIP network to common attacks.Security tools that are commonly used to address potential threats in VoIP networks are: the insertion of firewalls, the separation of data and voice networks and the adoption of access control lists. The introduction of such elements in the network can lead to unpredictable modifications on the performance of the network and, consequently, of the VoIP service. The paper presents the design and implementation of a reconfigurable test-bed for real-time measurements on VoIP systems, which provides the telecommunication engineer with means to plan the necessary changes in order to reach a higher level of security. In its basic version, the test-bed permits to take into account the right balance between security tools and the required Quality of Service (QoS)/Quality of Experience (QoE) for real time operations. An interesting feature of the enhanced version is the possibility of evaluating QoS and QoE in the presence of firewalls and different security configurations, and assessing the effect of the joint variation of QoS parameters on QoE results.

The Affects of Different Queuing Algorithms within the Router on QoS VoIP application Using OPNET

Voice over Internet Protocol (VoIP) is a service from the internet services that allows users to communicate with each other. Quality of Service (QoS) is very sensitive to delay so that VOIP needs it. The objective of this research is study the effect of different queuing algorithms within the router on VoIP QoS. In this work, simulation tool “OPNET Modeler version 14.0” is used to implement the proposed network (VoIP Network). The proposed network is a private network for a company that has two locations located at two different countries around the world in order to simulate the communications within the same location as a local and the communications between two locations as a long distance and analyze VoIP QoS through measuring the major factors that affect the QoS for VoIP according to international telecommunication union (ITU) standards such as: delay, jitter and packet loss. In this research , a comparison was carried out between different queue algorithms like First in First out (FIFO), Priority queue (PQ) and Weight Fair Queuing (WFQ) and it was found that PQ and WFQ algorithms are the most appropriate to improve VoIP QoS .

Attack Determination and its Security Analysis for Personal Communication in VoIP Networks

Voice over Internet Protocol (VoIP) is a technology that enables one to make and receive calls through the Internet instead of using the traditional analog PSTN (Public Switched Telephone Network) lines. Gtalk and Skype are frequently used for this purpose, which is a third party. So, we are utilizing their environment for conferencing. We are in need of a server to transfer the information from source to destination i.e. we are compromising our privacy at a certain level as they do not provide full security to our audio packets. Here, we do propose, novel flow analysis attacks that ensures both privacy and demonstrates the vulnerabilities in peer-to-peer VoIP networks. Solutions are proposed by quantifiable k-anonimity metrices.

Voice Priority Queue Scheduling System Models for VoIP over WLANs

The Voice over Internet Protocol (VoIP) is a delay sensitive traffic due to real-time applications on networks. The assessment of voice flow quality in the VoIP is an essential requirement for technical and commercial motivation. The packets of VoIP streaming may experience drops because of the competition among the different kinds of traffic flow over the network. A VoIP application is also sensitive to delay and requires the voice packets to arrive on time from the sender to the receiver side without any delay over WLAN. The scheduling system model for VoIP traffic is an unresolved problem. In this research paper, the author proposes a new Voice Priority Queue (VPQ) scheduling system models and algorithms for the VoIP over WLANs to solve scheduling issues over IP-based networks. They present new contributions, through the three stages of the VPQ. The VPQ scheduling algorithm is provided as an essential technique in the VoIP communication networks to guarantee the QoS requirements. The design of the VPQ is managed by the limited bandwidth utilization and has been proven to have an efficient performance over WLANs.

Evaluating the perceived voice quality on VoIP network using interpolated FIR filter algorithm

Voice over Internet Protocol (VoIP) is a popular communication service nowadays. VoIP reduces the cost of call transmission by passing voice and video packets through the available bandwidth for data packets through Internet protocol. The quality of the VoIP signal is degraded due to the various network impairments. The proposed scheme, interpolated finite impulse response, is implemented as post-processor after decoding the signal in VoIP system. The performance of the proposed scheme is evaluated for various network conditions. The results of the proposed scheme are measured with the objective measurement methods for signal quality evaluation. The performance of the proposed system is compared with the existing techniques for quality improvement in VoIP system. The results show much improvement in speech quality with the proposed scheme in comparison to other similar schemes.

A dynamic traffic allocation model of voice over internet protocol and public switched telephone network

PurposeCorporations with multiple national and international sites rely on telecommunication services such as public switched telephone network (PSTN) to deal with their customers and with vendors. Over the last few years, the cost of using PSTN for telecommunication has increased, thus adding considerable cost to business. The infrastructure of the internet is well placed now, and the continued growth of information technologies have made it cost effective to route voice calls over the internet, known as voice over internet protocol (VoIP). The purpose of this paper is to advance research, strategies and a predictive decision model that analyzes the cost of routing voice‐based traffic either via VoIP network or PSTN. This predictive decision model is based on the beta‐binomial distribution, and it computes the cost differential in tariff with respect to choice of routing the voice traffic between the VoIP network and the PSTN.Design/methodology/approachA call supported by the PSTN involves the establishment of an end‐to‐end circuit that is maintained for the duration of the call. A call supported by VoIP technology, by contrast, involves the transmission of many individual packets over an IP network. The cost of a VoIP call thus depends in part on the number and size of the packets that must be transmitted, i.e. the bandwidth required.FindingsGreat savings are realized when optimal number of calls would be routed through CCS vendors, as determined by the VoIP allocation model. The available bandwidth on the internet is capable of supporting VoIP at much cheaper rate than the expensive PSTN. Business could still buy into PSTN service if they require secured telecommunication services.Originality/valueThe paper tackles one of the most critical problems of minimizing the burden and costs of telecommunications by PSTN and VoIP.

Investigation of Automatic Speech Recognition Performance and Mean Opinion Scores for Different Standard Speech and Audio Codecs

Usage of Automatic Speech Recognition (ASR) systems is increasing day-by-day for voice centric applications in mobile handheld and Voice over Internet Protocol (VoIP) devices. The necessity is also increasing to find out the ASR performance under different network impediments. Among them, speech and audio coding standards is the one, which affects the ASR performance greatly, when, using them with different sampling and bit rates in the practical systems. Another common impediment which influences the ASR accuracy is the bit errors in the wireless networks and packet drop conditions in the VoIP networks. ASR performance with some of the speech coding standards under noise conditions for the wireless networks is reported in the literature. However, each study is reporting the ASR performance for few narrowband codecs with different speech databases and different ASR toolkits like RAPHEL, HTK, SPHINX, etc. In this paper, the analysis on ASR performance while using both narrowband and wideband speech and audio coding standards, which are currently accepted for GSM mobile and VoIP networks, using the common speech database-TIMIT, and using ASR toolkit-SPHINX, is presented. The Mean Opinion Score (MOS), which is the generally accepted speech quality measurement technique, is also analyzed for all the speech and audio coding standards, using the same speech database. The results of the studies carried out for the ASR word accuracies and MOS values for different narrowband and wideband speech and audio codecs under no-loss conditions are presented. Results for different rates of packet drop condition which is the common noise scenario in wired networks such as VoIP (which is also merging with wireless networks) are also presented. The observation is that though some of the codecs are showing poor MOS performance at lower bit rates, the corresponding ASR performance is comparable with other codecs at higher bit rates.

Mobility Impact on the End-to-End Delay Performance for VoIP over LTE

Long Term Evolution (LTE) is the last step towards the 4th generation of cellular networks. This revolution is necessitated by the unceasing increase in demand for high speed connection on LTE networks. This paper focuses on the performance evaluation of End-to-End delay under variable mobility speed for VoIP (Voice over IP) in the LTE network. In the course of E2E performance evaluation, realizing simulation approach three scenarios have been modeled using OPNET 16.0. The first one is the baseline network while among other two, one consists of VoIP traffic solely and the other consists of FTP along with VoIP. E2E delay has been measured for both scenarios in various cases under the varying mobility speed of the node. Simulation results have been studied and presented in terms of comparative performance analysis of the three network scenarios. In light of the result analysis, the performance quality of a VoIP network (with and without the presence of additional network traffic) in LTE has been determined and discussed. The simulation results for baseline VoIP network (non-congested) congested VoIP network and congested VoIP with FTP network show that as the speed of node is gradually increased, E2E delay slightly increases.