VoIP Users Research Articles

Supporting VoIP communication in IEEE 802.11ax networks: A new admission control and scheduling resource allocation scheme

The current IEEE 802.11ax standard enhances Wi-Fi networks with a series of new features, such as multi-user (MU) transmission, an Orthogonal Frequency Division Multiple Access (OFDMA) scheme and the ability to multiplex traffic from different access categories (ACs). These features can be utilized to enhance the QoS support for VoIP traffic and optimize the usage of IEEE 802.11ax network resources. This work proposes a new scheme to multiplex VoIP calls in IEEE 802.11ax MU frames and a new scheduler and resource allocation algorithm specifically designed for VoIP data traffic. Our scheduler allocates VoIP packets requiring longer transmission times into the same frame(s), minimizing the channel air-time assigned to VoIP transmission. In addition, unused radio resources in the MU frame are leveraged to transmit best-effort packets along with VoIP packets. For completeness, we also define a call admission control (CAC) algorithm that anticipates channel saturation conditions to ensure VoIP users can maintain a guaranteed level of QoS. Based on simulation results, our scheme is more efficient in reducing channel utilization than other schedulers such as multi-user round-robin (RR) (implemented by ns-3) or single-user FIFO. For example, for 30 VoIP stations using the G.711 codec under mixed channel conditions, our scheme reduces by 30% the air-time required to transmit VoIP packets. When coupled with the ability to also send best-effort packets along with VoIP packets, this translates into a higher throughput (i.e. 10 Mbit/s vs 4 Mbit/s) and more simultaneous VoIP users with guaranteed QoS (up to 46 VoIP users vs 26 and 28 users for the multi-user RR and single-user FIFO scheduling algorithms, respectively).

Toward a comprehensive subjective evaluation of VoIP users’ quality of experience (QoE): a case study on Persian language

Toward a comprehensive subjective evaluation of VoIP users’ quality of experience (QoE): a case study on Persian language

A Virtual Switching Model for Improvement of Call Quality in Integrated Communication System

Poor voice quality in VoIP models during communication has been a common occurrence which VoIP users experience, this can be frustrating when users cannot communicate efficiently. Most people find it difficult to think straight when they make calls and there is an echo. In addition to this frustration, the caller’s money, time, effort, energy is all wasted without compensation of any kind. Users are also frustrated by not receiving, transmitting or misunderstanding voice messages correctly. Given the need for voice quality in calls, it is of no importance when there is no proper communication. This study aims to reduce the threat of bad calls and improve the quality of voice calls. Nonetheless, we need to raise the filter duration to a high value in some real telecom’s environments with long echo delays. But, because of high computational complexity, it is not efficient in efficiency. In this study, we suggest a solution that uses a computational formula to compensate long echo, delay, packet loss, jitter and noise. The model designed was developed using MATLAB 2019b. This approach demonstrated productivity in terms of both voice quality and system speed.

Mitigating Coordinated Call Attacks On VoIP Networks Using Hidden Markov Model

Abstract This paper presents a 2-tier scheme for mitigating coordinated call attacks on VoIP networks. Call interaction pattern was considered using talk and salient periods in a VoIP call conversation. At the first-tier, Short Term Energy algorithm was used for call interaction feature extraction and at the second-tier Hidden Markov Model was used for caller legitimacy recognition. Data of VoIP call conversations were collated and analyzed to extract distinctive features in VoIP call interaction pattern to ascertain the legitimacy of a caller against coordinated call attacker. The performance metrics that was used are; False Error Rate (FER), Specificity, Detection Accuracy and Throughput. Several experiments were conducted to see how effective the mitigating scheme is, as the scheme acts as a proxy server to Session Initiation Protocol (SIP) server. The experiments show that; when the VoIP server is under coordinated call attack without a mitigating scheme only 15.2% of legitimate VoIP users had access to the VoIP network and out of which about half of the legitimate users had their calls dropped before completion, while with the 2-tier mitigating scheme, when the VoIP server is under coordinated call attacks over 90.3% legitimate VoIP callers had their calls through to completion

Teletraffic Analysis for VoIP Services in WLAN Systems with Handoff Capabilities

In this work, a TDMA-based system is studied for VoIP services with and without mobility. The system is considered to be dedicated for voice-only services such as a WLAN environment under the PCF mode or a cellular system. In such environments, mobiles usually have mobility. This introduces the need to study the system when a handoff procedure is enabled. The teletraffic analysis considers call arrivals, call completions, and ON/OFF activity processes for individual VoIP sessions. The VoIP system is simulated in order to verify the accuracy of the analytical results. Since many papers published in the literature consider a fixed number of VoIP users in the system, this analysis can be useful to estimate the system's performance when the system is in statistical equilibrium. As an additional feature of this work, a simple fluid model to calculate the number of active and inactive users in a VoIP system with and without handoff capabilities is proposed and developed.

The impact of the superposition coding concept on admission control strategy in OFDMA-based network

Admission control is a key mechanism to manage the increasing number of the simultaneous demanding services, requiring a desired Quality of Service (QoS) in a spectrum efficient manner. To address this issue, we investigate in this work, the use of the superposition coding technique to increase the system capacity through multiuser diversity exploitation. We propose a novel joint admission control and superposition coding formalism based on different utility functions: Opportunistic (OPSC), Great Fairness (GFSC) and Proportional Fairness (PFSC). Simulation results show the superiority of our proposed approaches over other by providing higher mean of served VoIP users and higher throughput while maintaining an average VoIP packet transmission delay lower than 0.6 ms.

SIP-based VoIP anomaly detection engine using DTV and ONR

VoIP has gained more attention in recent years due to its advantage of cheap calls when compared to the existing PSTN network. The callers such as the advertiser, telemarketers, prank callers who make use of this VoIP for generating the anomaly calls and messages are characterised as SPIT. The previous work detects the spam caller after getting the feedback from the callee. The proposed technique detects the anomaly in the call pattern without user involvement, i.e., the pre-acceptance method. This SIP-based approach relies on DTV and ONR of the caller to detect the anomaly calls and block the spammer. The parameter call duration, call count with frequency and the unique partner of the caller are used to compute the direct trust value of VoIP user. The ONR depicts the user behaviour in the digital shopping. The online shopping behaviour of the sender insists on the ONR value. The aggregation algorithm uses the DTV and ONR to measure the global reputation of the caller. This calculated global reputation value detects the anomalies and segregate the non-legitimate user during call setup using the session initiation protocol. The proposed system detects the spammer without analysing the content, without getting feedback from the user and before connecting the call.

Performance Evaluation of Cognitive Radio VoIP Users in Fading Environment

In this paper we investigate three channel allocation algorithms for multi-channel cognitive radio networks: SNR-based allocation (SA), queue-based allocation (QA), and queue-based SNR-aware allocation (QSA). The efficiency of the proposed algorithms is investigated for VoIP cognitive radio users (CRs). The resource allocation in the SA and QA schemes is performed based on the channel quality status seen by each CR and its queue length, respectively. The channel allocation in the QSA scheme is based on both the queue length and the SNR status. The mean packet loss rate (PLR) as the main quality-of-service (QoS) parameter is scrutinized for each algorithm. Novel analytical structures together with new analyses are developed, by which the level of provisioning QoS of VoIP traffic in each of the proposed algorithms can be evaluated. Numerical results and comparisons are also presented. In particular, the QSA algorithm is shown to present the best QoS level and yield the least PLR among the three allocation schemes.

WE-MQS-VoIP Priority: An enhanced LTE Downlink Scheduler for voice services with the integration of VoIP priority mode

The Long Term Evolution (LTE) is a high data rates and fully All-IP network. It is developed to support well to multimedia services such as Video, VoIP, Gaming, etc. So that, the real-time services such as VoIP, video, etc. need to be optimized. Nevertheless, the deployment of such live stream services having many challenges. Scheduling and allocating radio resource are very important in LTE network, especially with multimedia services such as VoIP. When voice service transmitted over LTE network, it is affected by many network impairments where there are three main factors including packet loss, delay, and jitter. This study proposes a new scheduler which is based on VoIP priority mode,Wideband (WB) E-model, QoS- and Channel-Aware (called WE-MQS-VoIP Priority scheduler) for voice services in LTE downlink direction. The proposed scheduling scheme is built based on the WB E-model and Maximum Queue Size (MQS). In addition, we integrate the VoIP priority mode into our scheduling scheme. Since the proposed scheduler considers the VoIP priority mode and user perception, thus, it improves significantly the system performance. The results demonstrate that the proposed scheduler not only meets QoS demands of voice calls but also outperforms Modified Largest Weighted Delay First (M-LWDF) in terms of delay, Packet Loss Rate (PLR) for all number of user (NU) and excepting NU equals 30, respectively. For Fairness Index (FI), cell throughput, and Spectral Efficiency (SE), the difference among the packet schedulers is not significant. The performance evaluation is compared in terms of Delay, PLR, Throughput, FI, and SE.

Adaptive probe-based congestion-aware handover procedure using SIP protocol

Wireless technologies have evolved very rapidly in recent years. In the future, operators will need to enable users to use communication services independently of access technologies, so they will have to support seamless handovers in heterogeneous networks. In this paper we present a novel adaptive congestion aware SIP based procedure for handover in heterogeneous networks. In the proposed algorithm the handover decision is based in addition to signal strength, also on target network congestion status, which is tested during the conversation. As SIP protocol was used, the proposed procedure is independent of access technologies. For performance evaluation of proposed procedure we developed a purpose‑built simulation model. The results show that the use of the proposed adaptive procedure significantly improves the QoE of VoIP users, compared to reference scenario, in which only signal strength was used as the trigger for handover decision.

A New Wireless VoIP Signaling Device Supporting SIP and H.323 Protocols

Nowadays, VoIP is a technology with a great demand and wireless networks are increasingly deployed. Each of these has its own technology constraints. For VoIP, it is very important to take into consideration the need to provide a high quality service according to well-defined standard transmission (jitter, end-to-end delay, MOS, and packet loss). However, wireless networks (IEEE 802.11) are based on radio which undergoes a number of technical constraints to achieve theoretical transmission rates; among these constraints the number of users of the networks, the distance between the client and the access, and the amount of data transmitted point are included. In this term, a study is made by simulating wireless network in OPNET Modeler with a fairly large number of VoIPs (15 users) whose signaling is handled via a new node that was created specifically to manage the signaling tasks under SIP and H.323 in order to minimize the number of nodes in the network and avoid the congestion. In this paper, two scenarios are compared; the first contains a number of VoIP users with SIP and H.323 signaling handled by the new created device; the second scenario is similar to the first except that the distance between the stations is remarkably lower.

An anomaly-based approach to the analysis of the social behavior of VoIP users

In this paper we present the results of a study we recently conducted by analyzing a large data set of VoIP Call Detail Records (CDRs), provided by an Italian telecom operator. The objectives of this study were twofold: (i) first, to provide a representation of users behavior, as well as of their mutual interaction and communication patterns, allowing to identify certain easily separable user categories; and (ii) second, to design and implement a framework calculating such a representation starting from CDR, capable of operating within certain time constraints, and grouping users using unsupervised techniques.The paper shows how we can reliably identify behavioral patterns associated with the most common anomalous behaviors of VoIP users. It also exploits the expressive power of relational graphs in order to both validate the results of the unsupervised analysis and ease their interpretation by human operators.

UTILITY BASED SCHEDULING AND CALL ADMISSION CONTROL FOR LONG TERM EVOLUTION NETWORKS

In this study, we propose to design a call admission control algorithm which schedules the channels for Real time and non-real time users. In Long Term Evolution (LTE) 3GPP Networks, several works were done on call admission control but these works rarely considers scheduling of resources to the real time and non-real time users.When the system meets traffic oriented performance degration, maximum resources are utilized for load balancing and to maintain the consistent quality. In order to avoid the channel degradation and improve the Quality of Service (QoS), the call requests are classified into New Call (NC) request and Handoff Call (HC) request and the type of services are classified as VoIP and video. Then based upon the Received Signal Strength (RSS) value, the channel is estimated as good channel or bad channel. Resource allocation is made for VoIP users based on traffic density. Then non-VoIP users and the non-real time users are allocated resource blocks using the channel condition based marginal utility function. When there are no sufficient resources to allocate, it allocates the resources of bad channel users there by degrading their service. We have designed the network topology with G (n) and B (n) for representing the available good and bad channels. We investigate the performance degradation when the real time, Non real Time, video and VOIP environments based on RSS threshold value.Comparison is made with the VOS in terms of the paramenters like throughput,bandwidth,delay,fairness and rate. Our proposed method provides good performance and quality.From our simulation results we show that this admission control algorithm provides channel quality and prioritizes the handover calls over new calls which allocates resources to all kinds of users.

Formal analysis for robust anti-SPIT protection using model checking

Anti-SPIT policies counter the SPam over Internet Telephony (SPIT) by distinguishing bots launching unsolicited bulks of VoIP calls from human beings. We propose an Anti-SPIT Policy Management mechanism (aSPM) that detects spam calls and prevents VoIP session establishment by the Session Initiation Protocol (SIP). The SPIN model checker is used to formally model and analyze the robustness of the aSPM mechanism in execution scenarios with parallel SIP sessions. In case of a possible design flaw, the model checker provides a trace of the caught unexpected behavior (counterexample), that can be used for the revision of the mechanism’s design. Our SPIN model is parameterized, based on measurements from experiments with VoIP users. Non-determinism plays a key role in representing all possible anti-SPIT policy decisions, in terms of the SIP messages that may be exchanged. The model checking results provide evidence for the timeliness of the parallel SIP sessions, the absence of deadlocks or livelocks, and the fairness for the VoIP service users. These findings ensure robust anti-SPIT protection, meaning that the aSPM mechanism operates as expected, despite the occurrence of random SPIT calls and communication error messages. To the best of our knowledge, this is the first analysis for exhaustively searching security policy flaws, due to complex interactions between anti-SPIT measures and the SIP protocol services.

Improving ertPS Grant Allocation for VoIP Traffic inSilence Duration

This paper proposes a new uplink scheduling algorithm that can increase the uplink VoIP user capacity for ertPS scheduling in 802.16e networks. This algorithm solves the problems of uplink resources wastes and large overhead which causes during silence period. We compare our method with four related schemes including conventional ertPS. In these algorithms, efficient schemes are proposed to improve the uplink resource allocation procedure for ertPS scheduling. We use OPNET simulator to investigate delay, jitter and loss in our experiments. We show that our scheme improves the capacity by 20% while the voice quality is affected minimally.

Utility Based Scheduling and Call Admission Control for LTE (3GPP) Networks

In this paper, we propose to design a call admission control algorithm which schedules the channels for real time and non-real time users based on Utility function. In Long Term Evolution (LTE) 3GPP Networks, several works were done on call admission control but these works rarely considers scheduling of resources to the real time and non-real time users. The call requests are classified into new call (NC) request and handoff call (HC) request and the type of services are classified as VoIP and video. Then based upon the received signal strength (RSS) value, the channel is estimated as good channel or bad channel. Resource allocation is made for VoIP users based on traffic density. Then non-VoIP users and the non-real time users are allocated resource blocks using the channel condition based marginal utility function. When there are no sufficient resources to allocate, it allocates the resources of bad channel users there by degrading their service. Thus from our simulation results we show that this admission control algorithm provides channel quality and prioritizes the handover calls over new calls which allocates resources to all kinds of users.

Smart Phone VoIP 서비스에 대한 공격과 도청 연구

VoIP service by taking advantage of the current PSTN network and internet over the existing telephone network at an affordable price allows you to make voice calls to the service is being expanded. However, the security of public must be maintained for security vulnerabilities in Smart Phone VoIP case problems arise, and is likely to be attacked by hackers. In this paper, the Internet, using wired and Smart Phone VoIP services may occur during analysis of the type of incident and vulnerability analysis, the eavesdropping should conduct an attack. Smart Phone VoIP with institutional administration to analyze the vulnerability OmniPeek, AirPcap the equipment is installed in a lab environment to conduct eavesdropping attack. Packet according to the analysis and eavesdropping attacks, IP confirmed that the incident as an attack by the eavesdropping as to become the test proves. In this paper, as well as Smart Phone VoIP users, the current administration and the introduction of Smart Phone service and VoIP service as a basis for enhanced security will be provided.

The Study on Integrated Network Architecture in Next Generation Network Services

Next generation networks supporting multimedia applications include hybrid/integrated satellite-terrestrial network architectures require stringent QOS provisioning for fixed and/or mobile services. In this paper, we have described Integrated Network architecture and technical challenges for the system realization based on the ITU-R current work plan. We developed a cross-layer design based simulation framework in order to understand the impact of rain fade and mitigation due to DVB-S2 on the multimedia applications. our simulation network models include DVB-S2/RCS for the satellite segment and two scenarios for terrestrial segments namely WiMAX and WiFi. In particular, our simulations demonstrated the advantage of cross layer design in system capacity gains for VoIP users in hybrid network with WiMAX as terrestrial segment. Also our results show the impact of fading on applications such as video conferencing VoIP, HTTP, Data Base Query and FTP in terms of QOS parameters e.g. bandwidth allocation, end-to-end delay, delay variation.

A novel SIP based procedure for congestion aware handover in heterogeneous networks

One of the next steps in the current rapid evolution of wireless technologies will be that operators will need to enable users to use communication’ services independently of access technologies, so they will have to support seamless handovers in heterogeneous networks. In this paper we present a novel SIP based procedure for congestion aware handover in heterogeneous networks. With newly defined SIP messages the handover decision is based not only on signal strength, but also on target network status. Using the SIP protocol, this approach is completely independent of access technology and the underlying protocol used, and thus can be easily deployed in operator’s environments. The proposed procedure was evaluated with a purpose-built simulation model. The results show that, using the proposed procedure, the QoE of VoIP users can be maintained during session by eliminating handovers to target network, which could cause degradation of service in the target network and by triggering handovers to another network, when there could come to degradation of service in the current network.

Improving wireless VoIP quality by using adaptive packet coding

The transfer of real-time VoIP data over wireless and mobile networks is subject to packet losses that yield unacceptable quality. This work proposes that the quality can vary significantly depending on the packet coding method used. An adaptive piggyback packet coding that can provide the VoIP user acceptable quality over the wide range of packet loss conditions observed in wireless and mobile networks is found.