VoIP Networks Research Articles

New Improved echo canceller based on Normalized LMS Adaptive filter for Single talk and Double talk Detection, Subband echo cancellation, Acoustic Echo cancellation

A modified version of normalized least mean square (NLMS) algorithm that achieves faster convergence in time domain with a marginal increment in implementation complexity than the existing one is proposed. Variations on the Normalized LMS (NLMS) algorithm have been made to improve convergence rate, Signal to Noise Ratio (SNR) and Steady state characteristics for Single talk and Double talk detection, Sub band Echo cancellation and Acoustic echo cancellation. The echo canceller is intended to cancel wireline echo from the PSTN due to the 2-to-4 wire hybrid conversion impedance mismatch. The echo canceller will also cope with acoustic echo originating from the wireline phone as well as analog cellular hands-free calls in the near-end path. Traditionally, speech enhancement techniques in VOIP or mobile networks are done in the transcoding unit operating on the uncoded signal. This means that the signal coming from the mobile terminals for example has to be decoded, enhanced and encoded again; these operations obviously introduce delays other than being computationally intensive and particularly prone to adding further quantization noise. Hence the need to design a echo canceller that offers a better convergence rate and Steady state characteristics.

A Garch-based adaptive playout delay algorithm for VoIP

Network delay, packet loss and network delay variability (jitter) are important factors that impact on perceived voice quality in VoIP networks. An adaptive playout buffer is used in a VoIP terminal to overcome jitter. Such a buffer-control must operate a trade-off between the buffer-induced delay and any additional packet loss rate. In this paper, a Garch-based adaptive playout algorithm is proposed which is capable of operating in both inter-talkspurt and intra-talkspurt modes. The proposed new model is based on a Garch model approach; an ARMA model is used to model changes in the mean and the variance. In addition, a parameter estimation procedure is proposed, termed Direct Garch whose cost function is designed to implement a desired packet loss rate whilst minimising the probability of consecutive packet losses occurring. Simulations were carried out to evaluate the performance of the proposed algorithm using recorded VoIP traces. The main result is as follows; given a target Packet Loss Rate (PLR) the Direct Garch algorithm produces parameter estimates which result in a PLR closer than other algorithms. In addition, the proposed Direct Garch algorithm offers the best trade-off between additional buffering delay and Packet Loss Rate (PLR) compared with other traditional algorithms.

Communications on the Web

User experience and the resulting economics dictate the migration of real time communications such as voice and video from dedicated VoIP networks to the Web and to the full integration with other rich Internet applications. This leads to adopting the Web architecture and protocols and security and renouncing various network application protocols such as SIP and XMPP. The key objective is to make communications accessible to millions of Web developers and enable Web sites with communications. This will be a transformational development for telecommunications, to be gradually absorbed by the Internet.

VoIP network architectures and impacts on costing

PurposeThis paper aims to describe the effect of VoIP network architectures on the cost modelling of termination rates of VoIP services.Design/methodology/approachThe study investigates and organises the arguments available in the technical and regulatory field related to VoIP networks and services in order to ascertain the possible impact of VoIP techniques, the provisioning of voice features in VoIP networks, and network interconnection issues on the cost of regulated VoIP services.FindingsThe information and analysis reveals how the provision of VoIP services is related to a number of issues that will have an effect on the cost of VoIP termination rates. In particular, the study analyses the impact on a cost model of the components of a VoIP network architecture, the usage factor of network elements, and the traffic volume generated by VoIP applications.Research limitations/implicationsThe issues described in the article can be used in elaborating a cost model for termination rates in VoIP networks. For the present study, no cost model was built, and therefore no quantitative estimations were made of the specific impact of every cost parameter on the termination rates.Practical implicationsThe findings of this study can be used by policy makers, voice operators, and researchers.Originality/valueMost studies of VoIP that are available in the literature address, on the one hand, the costs of corporate VoIP networks and, on the other, the regulation of VoIP services. This article, however, presents a comprehensive study of the most relevant features of VoIP network architectures that should be considered when determining regulated termination rates.

Reward mechanisms for P2P VoIP networks

As we have seen, P2P VoIP software, such as Skype, has emerged from the current generation of telecommunication systems. However, establishing communication applications based on P2P networks without considering the operational models of the Internet presents potential dangers. In this study, we treat a VoIP telephone conversation as a dynamic game and compare the ex post reward mechanism with the ex ante reward mechanism in forward versus discard and QoS routing. Our simulation results point out that the ex post reward mechanism is better than the ex ante reward mechanism in forward versus discard; however, the opposite holds true in QoS routing when the expected number of periods is sufficiently large. In addition, this study considers the reward mechanisms and the platform provider's benefit to find the optimal number of supernodes in a transmission path.

A secure and efficient SIP authentication scheme for converged VoIP networks

Session Initiation Protocol (SIP) has been widely used in current Internet protocols such as Hyper Text Transport Protocol (HTTP) and Simple Mail Transport Protocol (SMTP). SIP is a powerful signaling protocol that controls communications on the Internet for establishing, maintaining and terminating sessions. The services that are enabled by SIP are equally applicable to mobile and ubiquitous computing. This paper demonstrates that recently proposed SIP authentication schemes are insecure against attacks such as off-line password guessing attacks, Denning-Sacco attacks and stolen-verifier attacks. In order to overcome such security problems, a new secure and efficient SIP authentication scheme in a converged VoIP network based on elliptic curve cryptography (ECC) is proposed and it works to exploit the key block size, speed, and security jointly.

Performance analysis of non‐preemptive hol GE/G/1 queue with two priority classes of sip‐t signaling system in carrier grade VoIP network

For analytic simplicity, the priority queueing of a VoIP network is modeled under the assumption that the traffic arrival process is Poisson distributed and the service time is exponentially distributed, while knowing that the holding times of calls are not an exponential distribution. Under this condition, the non‐Markovian process (e.g. Supplementary Variable Technique) is often considered. However, the analysis for priority processes has always been difficult to tackle directly, particularly when the number of priorities is greater than two. Earlier studies were therefore restricted to two priority classes (e.g. M1 B1, M2 B2/G1, G2/1). Although the analytical results are available, very complicated closed‐form expressions of mean queue length are still required for each class. The objective of this paper is to obtain a simple and analytic closed‐form for the mean length of an HOL (Head of Line) GE/G/1 queue with more than two priority classes. Compared with M 1 B1, M2 B2/G1, G2/1, the HOL GE/G/1 priority queue computations of an SIP‐T signaling system require only a few parameters (e.g. mean and variance) since measurements of actual interarrival or service times are often constrained. The performance analyses from mathematical calculations and computer simulations further confirm the accuracy and robustness of an HOL GE/G/1 queue. Hence, the optimum values of interarrival time SQVs (Squared Coefficient of Variations) and service time SQVs can be determined for GM (Guaranteed SIP‐T messages) and NGM (Non‐guaranteed SIP‐T messages).

Development of an innovative VoIP Student Attendance Recording Application

This project aims to provide an in-depth technical in-sight into development of an innovative student attendance recording application using emerging communication technologies. Development of the Voice-over-IP Student Attendance Recording Application (VoIP SARA) for the Box Hill Institute of TAFE (BHI) is presented. This project demonstrates application programming, test bed setup, design considerations, design decisions and future plans, as well as creative usage of emerging communications technologies. The developed application combines Cisco VoIP telephony and network systems, and integrates it with existing database systems with the aim to provide BHI’s management with a cost effective, efficient and secure way to record and retrieve student attendance data. This project is a success and when fully integrated within the existing system, it will provide BHI with a cost effective and efficient student attendance recording and reporting system. We conclude that this application has a potential for use in recording and retrieving student attendance at other Australian education providers.

Echo cancellation in VoIP with improved least square lattice method

In this work, an approach to echo cancellation in VoIP network based on the packet-based echo cancellation model was employed. In addition, a comparative validation was performed and then analysed specifically in the state of echo canceller as adaptive filtering was done by using three algorithms: normalised least mean square (NLMS), recursive least square (RLS) and least square lattice (LSL). Communication environments with three different echo strength levels: 20%, 50% and 70% of far-end speech were simulated. The result shows that in terms of time computation, NLMS is the best, followed by LSL and RLS as the last. However, NLMS suffers from the measure of ERLE, as it falls to the third place, followed by LSL and RLS as the first. Then, taking them all into consideration, it could be concluded that LSL is the best among all for echo cancellation as verified in the experimental simulation.

Improving Distributed Denial of Service (DDOS) Attack Detection Performance using Priority Queuing

Distributed Denial of Service (DDoS) attacks are a major threat for the deployment of SIP-based VoIP networks. Various solutions have been suggested to mitigate the impact of these attacks, each bringing its advantages and disadvantages. Classifying the SIP-based VoIP traffic into different queues according to the suspiciousness of the traffic, can be an efficient solution if it is correctly implemented. Unfortu-nately, adding queues to the SIP servers affects severely its performance. The current paper suggests a mechanism called “virtual queuing” that implements a priority queue on a SIP server while keeping its performance at its best. Keywords: DDoS attacks, Prioritization, Queuing, Scheduling Algorithm.

Bandwidth calculation for VoIP networks based on PSTN statistical model

This paper shows an analysis how to calculate proper bandwidth for VoIP calls after proper dimensioning of PSTN network. For this purpose, we use Erlang B and extended Erlang B formulae. Further, we have developed a software tool, named Bandwidth Calculator to calculate proper number of the circuits on the PSTN side and after that IP bandwidth. Traffic analysis is conducted for VoIP networks considering impact of many factors on the bandwidth such as: voice codecs, samples, VAD, RTP compression. The results obtained by bandwidth calculator are compared to simulation results and data obtained by measurements. .

A Policy-Based Admission Control Scheme for Voice over IP Networks

Problem statement: In Voice Over IP (VOIP) network, when more calls are admitted to the network, more voice packet traffic is created. Since bandwidth is always limited, this may result network congestion and/or may affect voice quality. Thus, we needed a mechanism for improving the Quality of Service (QoS) by controlling VOIP calls admission. Approach: Given a specified bandwidth and a constant background data rate, we attempted to explore the effect of Open Window and Leaky Bucket admission schemes on VoIP calls quality. These policy-based admission controls were simulated using NS-2 Simulator. The inter-arrival time distribution for the network background data traffic was assumed to be deterministic with a Constant Bit Rate (CBR). Voice packets traffic inter-arrival time is assumed to have an exponential distribution. Each voice call has a rate of 64 kbps for duration of 120 min. Results: Various performance measures of VoIP calls and packet traffic were evaluated including: packet loss, packet drop rate, delay, jitter and call rejection rate. Performance results of the experiment are summarized in a power ratio index which presented the impact of a collection of performance parameters on VoIP service quality. Conclusion: Implementing a policy based admission scheme on VoIP network will improve its QoS and the degree of improvement depends on the network setting parameters. If threshold rate for call admission is set above network ceiling bandwidth, leaky bucket will result a higher and unacceptable jitter. Overall, leaky bucket scheme was considered inferior when compared to open window for improving QoS of VoIP.

Survey of network security systems to counter SIP-based denial-of-service attacks

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 open flooding attacks issues could be addressed.

Performance Analysis of Transmitting Voice over Communication Links Implementing IPsec

The prevalence and ease of packet sniffing and other techniques for capturing packets on an IP based network makes encryption a necessity for VOIP(Voice Over Internet Protocol). Security in VOIP is concerned both with protecting what a person says(by Encryption) as well as to whom the person is speaking(by Authentication). IPsec can be used to achieve both of these goals as long as it is applied with ESP(Encapsulating Security Payload) using the tunnel method. This secures the identities of both the endpoints and protects the voice data from prohibited users once packets leave the corporate intranet. The incorporation of IPsec with IPv4 increase the availability of encryption, VOIPsec (VOIP using IPsec) helps reduce the threat of man in the middle attacks, packet sniffers, and many types of voice traffic analysis. Combined with the firewall implementations, IPsec makes VOIP more secure than a standard phone line, In this paper the negative effect of adding security to VoIP networks has been measured for different simulation times using OPNET(Network Simulation Tool). the results show that transmitting voice over IPsec increase the end to end delay,delay variation(jitter),packet loss and call setup time.

Influence of informational content of background noise on speech quality evaluation for VoIP application

With the rise of mobile communication systems, the background noise in the speaker's environment and its interaction with VoIP network impairment affects speech quality perception. This effect should be taken into account in nonintrusive models in order to improve accuracy of end user perception measurement. The aim of this study is to determine the impact of information contained in background noise (background speech, environmental sources) on speech quality perception. A subjective test on the speech quality perception in real network conditions has been done comparing the effect of stationary background noise mainly due to transmission equipment (electrical humming or blowing noises) with the effect of nonstationary environmental noise (public place, traffic noise, background conversation). Interactions between these different background noise condition and realistic network impairments (coders and packets loss) were also studied. The speech quality was evaluated through the Mean Opinion Score determined from an Absolute Category Rating method.

Online pairing of VoIP conversations

This paper answers the following question; given a multiplicity of evolving 1-way conversations, can a machine or an algorithm discern the conversational pairs in an online fashion, without understanding the content of the communications? Our analysis indicates that this is possible, and can be achieved just by exploiting the temporal dynamics inherent in a conversation. We also show that our findings are applicable for anonymous and encrypted conversations over VoIP networks. We achieve this by exploiting the aperiodic inter-departure time of VoIP packets, hence trivializing each VoIP stream into a binary time-series, indicating the voice activity of each stream. We propose effective techniques that progressively pair conversing parties with high accuracy and in a limited amount of time. Our findings are verified empirically on a dataset consisting of 1,000 conversations. We obtain very high pairing accuracy that reaches 97% after 5 min of voice conversations. Using a modeling approach we also demonstrate analytically that our result can be extended over an unlimited number of conversations.

A novel design of a VoIP firewall proxy to mitigate SIP-based flooding attacks

This paper proposes a novel method to address the protection necessary to mitigate flooding attacks in VoIP networks which can produce rapid saturation of a firewall and crippling of a VoIP switch. The paper proposes a stateless firewall nonce checking mechanism as an extension to the existing (stateful) SIP digest authentication. This combination aims to form a more secure and flood-resistant authentication scheme for SIP-based VoIP systems. The proposed mechanism has been implemented on a Linux iptables firewall and the experimental results demonstrate proof-of-concept showing that by incorporating this mechanism it is possible to provide substantially improved SIP-based flooding mitigation.

Strategies to Keep Your VoIP Network Secure

As VoIP technology matures, more IT departments have made the jump to integrate VoIP into their enterprise communication systems. But, before an organization fully commits to the technology, it should perform a risk assessment, paying close attention to any security measures needed to protect these IP-based voice networks. For telephone systems where controlled access throughout the communications infrastructure isn't necessary, services that leverage machines on the public Internet could be a cost-effective choice. However, for systems where secured network control is critical, VoIP networks based on private enterprise infrastructures should be considered. This article focuses on the security measures associated with private-enterprise-based VoIP networks.

Performance Enhancement on Voice using VAD Algorithm and Cepstral Analysis

The Transport of Voice Over Network has been existing for a long time. Due to the bad quality of speech and absence of useful service, it has not yet been widely spread. The benefits of reduced cost and bandwidth savings of carrying voice over data network is associated with Quality of Service (QoS). Many different techniques and protocols are used to improve the quality of service in voice over data network. We review our data on the voice quality effects of background noise and the compression option of suppressing transmission during silence. Because the main problem of echo has emerged repeatedly in the VOIP environment, we review this issue in voice over data communication network is used to VAD algorithms and cepstral analysis of VOIP network performance.

Packet based echo cancellation for VoIP networks

In today’s modern telephony network, VoIP is fast emerging as one of the main communication techniques. However, the performance and the quality of VoIP are affected by echo. Packet Based Echo Canceller (PBEC) is introduced, as a solution to cancel echo in the VoIP network. PBEC can replace the current echo cancellers, which are located in the Public Switched Telephony Network (PSTN) central switches. The operating principle of the PBEC is explained and its advantages are highlighted. The performance of the PBEC using different speech codecs is also studied. Using the PBEC, a maximum Echo Return Loss Enhancement (ERLE) of 37.39 dB has been achieved when used with the Pulse Code Modulation (PCM) based speech codec. From the simulation results, it can be seen that the performance of the Adaptive Differential Pulse Code Modulation (ADPCM) clearly matches the performance of the PCM based speech codec. The other major problem affecting the VoIP network is the issue of packet loss. This issue of packet loss has been successfully addressed in this paper by the insertion of random values. With the insertion of random values, the ERLE increases by 4.81 dB compared to when there is no insertion of random value. The PBEC with the utilization of random values would make the VoIP a better communication tool.