VoIP System Research Articles

A Fuzzy Logic Based Method for Analysing Test Results

Network operators must perform many tasks to ensure smooth operation of the network, such as planning, monitoring, etc. Among those tasks, regular testing of network performance, network errors and troubleshooting is very important. Meaningful test results will allow the operators to evaluate network performanceof any shortcomings and to better plan for network upgrade. Due to the diverse and mainly unquantifiable nature of network testing results, there is a needs to develop a method for systematically and rigorously analysing these results.In this paper, we present STAM (System Test-result Analysis Method) which employs a bottom-up hierarchical processing approach using Fuzzy logic. STAM is capable of combining all test results into a quantitative description of the network performance in terms of network stability, the significance of various network erros, performance of each function blocks within the network. The validity of this method has been successfully demonstrated in assisting the testing of a VoIP system at the Research Instiute of Post and Telecoms in Vietnam.The paper is organized as follows. The first section gives an overview of fuzzy logic theory the concepts of which will be used in the development of STAM. The next section describes STAM. The last section, demonstrating STAM’s capability, presents a success story in which STAM is successfully applied.

Unjuk Kerja IP PBX Asterisk dan FreeSWITCH pada Topologi Bertingkat di Jaringan Kampus

Internet Protocol-based Private Branch Exchange (IP PBX) is needed to interconnect/switch calls between VoIP communication devices in a Local Area Network (LAN). VoIP service can be provided by using IP PBX in data/IP networks that are already provisioned. We conduct this research to evaluate VoIP performances of two IP PBXs using Asterisk and FreeSWITCH. We design an experimental network architecture that is adapted from the hierarchical topology in a Campus Area Network (CAN) at the research location that consists of core switch, distribution switch, and access switch. The two IP PBXs are placed at different locations in the hierarchy of the network topology for investigating the effects of the placement to the VoIP performance. This experimental concept can be used as a base for designing architectures of centralized and distributed IP PBXs in campus network. We measure VoIP performance by evaluating the systems with the Quality of Service (QoS) and the Quality of Experience (QoE) parameters. We also monitor processor and memory consumptions that are used by each VoIP server software when communication takes place. The evaluation was conducted on VoIP system with phone-to-phone scenario in the experimental network that is connected to the active campus network. Our results show that IP PBX placement at different hierarchy of the network topology affects to the VoIP performance particularly on Asterisk IP PBX. The performance of FreeSWITCH IP PBX in terms of QoS and QoE is slightly better than the Asterisk one on average latency/delay, jitter, Mean Opinion Score (MOS), and memory consumption. The values are 41,012 ms, 0,060 ms, 5, 1 % for IP PBX 1, and 41,016 ms, 0,066 ms, 5, 0,5 % for IP PBX 2.

Wiretapping End-to-End Encrypted VoIP Calls: Real-World Attacks on ZRTP

AbstractVoice calls are still one of the most common use cases for smartphones. Often, sensitive personal information but also confidential business information is shared. End-to-end security is required to protect against wiretapping of voice calls. For such real-time communication, the ZRTP key-agreement protocol has been proposed. By verbally comparing a small number of on-screen characters or words, called Short Authentication Strings, the participants can be sure that no one is wiretapping the call. Since 2011, ZRTP is an IETF standard implemented in several VoIP clients.In this paper, we analyzed attacks on real-world VoIP systems, in particular those implementing the ZRTP standard. We evaluate the protocol compliance, error handling, and user interfaces of the most common ZRTP-capable VoIP clients. Our extensive analysis uncovered a critical vulnerability that allows wiretapping even though Short Authentication Strings are compared correctly. We discuss shortcomings in the clients’ error handling and design of security indicators potentially leading to insecure connections.

Performance Analysis of Data Transmission over Wireless VOIP using Wavelets

Objectives: To analyze the performance of wireless VOIP system using wavelets. Method: In this paper a Wavelet based Data (Image, Audio and Video) Transmission System for wireless VOIP is proposed. The performance of each wavelet in this proposed system is analyzed on the basis of SNR, PSNR, RMSE, PRSE and Compression ratio. Packet Loss and Delay is also evaluated to analyze transmission capabilities. All these parameters are evaluated for the proposed system without using wavelets and with using different wavelets. Findings: The simulation results shows that the proposed system gives enhanced performance by using different wavelets when compared by without using wavelets. Finally, the Daubechies family has shown better results when compared with its counterpart’s. Applications: This wavelet based proposed system can used for high speed 4G / 5G IP based systems.

Novel DTD and VAD assisted voice detection algorithm for VoIP systems

Echo cancellation plays an important role in current Internet protocol (IP) based voice interactive systems. Voice state detection is an essential part in echo cancellation. It mainly comprises two parts: double talk detection (DTD) and voice activity detection (VAD). DTD is used to detect doubletalk and prevent filter divergence in the presence of near-end speech, and VAD is used to determine the near-end voice activity and output silence indicator when near-end is silent. However, DTD straightforwardly proceeded may mistakenly declare double talk under double silent condition, coefficients update under the far-end silence condition may lead to filter divergence, and current VAD algorithms may misjudge the residual echo from the near end to be far-end voice. Therefore, a voice detection algorithm combining DTD and far-end VAD is proposed. DTD is implemented when VAD declares far-end speech, filtering and coefficients update will be halted when VAD declares far-end silence, and the far-end VAD adopted is multi-feature VAD based on short-time energy and correlation. The new algorithm can improve the accuracy of DTD, prevent filter divergence, and exclude the circumstance that far-end signal only contains residual echo from near end. Actual test results show that the voice state decision of the new algorithm is accurate, and the performance of echo cancellation is improved.

Speech Compression

Speech compression is a key technology underlying digital cellular communications, VoIP, voicemail, and voice response systems. We trace the evolution of speech coding based on the linear prediction model, highlight the key milestones in speech coding, and outline the structures of the most important speech coding standards. Current challenges, future research directions, fundamental limits on performance, and the critical open problem of speech coding for emergency first responders are all discussed.

Comparative Study and Analysis of various VoIP coding Algorithms

over internet protocol (VoIP) is a method of providing phone services over dedicated public IP networks. It allows significant cost savings over traditional Public Switched Telephone Networks (PSTN). Speech quality, as perceived by the users of VoIP telephony, is critically important. Signal quality of the VoIP system is degraded by various network layer problems, which include delay, packet loss and jitter. The implementation of signal through various coding algorithms and digital signal processor can improve the quality of degraded VoIP signal. The present work deals with comparative study and analysis of VoIP Codecs (G.711, G.729, AMR, AMR-WB etc.). The VoIP simulations are conducted for G.711, G.729 and AMR-WB speech coders for different network conditions. The coding algorithms are implemented on VoIP speech signal. The results are validated through the measurement of enhancement signal using perceptual evaluation of speech quality (PESQ)/MOS measurement.

VoIP-quality of experience modeling: E-model and simplified E-model enhancement using bias factor

The E-model is a non-intrusive measurement method that many researchers have applied to the study of VoIP quality measurement. While the Simplified E-model is a modified version from the original, it can still be used as an alternative solution. Nevertheless, it has been found that the E-model and the Simplified E-model still require further improvement. Therefore, to enhance the original E-model, this paper proposes a new factor. Moreover, the Simplified E-model has also been enhanced by the same approach. Based-on the Thai environment, the new factor called Thai Bias factor, can be computed by subtracting the subjective test results using conversation tests with native Thai users from the objective test results using an E-model tool and the Simplified E-model calculation. Of course, both E-mode tests and conversation tests were conducted with the same VoIP system and test scenarios. The Enhanced E-model and the Simplified E-model using the Thai Bias factor were then evaluated by comparing the test set from other groups of native Thai users. After evaluation of the improved models, it has been found that the Enhanced E-model and the enhanced Simplified E-model can gain higher confidence. The Enhanced E-model delivers improved accuracy and reliability at approximately more than 20 % when compared to an available E-model tool, while the Enhanced Simplified E-model delivers improved performance at approximately more than 46 % when compared to Simplified E-model calculation.

Extending SIP to support payments in a generic way

The growth of users connected to the Internet with a high bandwidth connection hasfavored the increase of multimedia services. As many of these services are provided by means of SIP, adding support for payment to SIP might benefit vendors. Payments in SIP have been proposed for accessing services, for microbilling and even as a solution to SPAM in VoIP systems. Current proposals have some limitations such as either not being suitable for low payments or micropayments, or not supporting the use of different payment protocols or the payment is always made for the whole session or they do not take into account that streams of different quality could have different prices. In response to these limitations, we propose a new SIP extension for supporting any kind of payments (both micropayments and macropayments) on SIP. In addition to being payment-independent, our proposal solves interoperability problems. Our proposal is based in an standard extension of SDP and SIP in order to maximize the compatibility. This facilitates the deployment of payment in SIP-based services. Moreover, our SIP extension supports an offer/answer model that allows the choice of the quality of the streams as well as the payment options to use. Furthermore, it is flexible and not only supports payment but also more complex business models such as loyalty models, credentials and subscriptions. In this paper, we provide a generic way to incorporate new payment methods and business models in the vendor's software. Using some application scenarios, we make a comparison between our proposal and previous work to show some of the advantages of using our proposal.

Study SIP Protocol on Asterisk Phone System and Offer Solutions to Its Security

Undoubted every organization’s heart is it’s phone system. The old phone systems couldn’t perform any method to make phone center and voice transmission intelligent on network and they had determinate abilities. Meantime Voice over Internet Protocol (VoIP) introduced itself to the world and performed a lot of abilities for clients like voice transmission on network. Many companies Investment on voip systems and implemented their methods on software and hardware packages. But between them a different production had designed and Performanced by Mark Spenser from Digium Company in 1992 which named Asterisk. Asterisk’s increasing popularity’s reason was its open code programs and its flexibility. VoIP systems such as Asterisk use voice transmission protocols to transfer voice over network. One of the voice transmission protocols is Session Initiation Protocol (SIP), which is one of the Asterisk’s voice transmission protocols. The first and the most important point in voice transmission over network is security. Security can be divided to two parts as inscrutability of invaders to network and coding transmitted voices over network to prevention of illegal listening. In this project at first we tried to introduce Asterisk phone system’s structure and Session Initiation Protocol (SIP) and then scrutiny method of Invader’s dominance to this protocol and Performance modern methods to prevent hacker’s dominance and also coding voice packages to Obscure them in transmission way.

Presenting Solutions to Increase Simultaneous Call in VOIP System by SIP Protocol - Based Media Server

By development of multi-media in networks, the borders among networks are changed and all networks are approaching to be united. Unity of data, video and voice networks in one network has many advantages and disadvantages for users and servers. One of the disadvantages is unwanted events in network including load increase, jitter, information packet loss, delay and etc. and all these lead into low quality of voice and disconnection during simultaneous call. The use of multimedia server is one of the efficient ways to improve VOIP. We can enable the video conferences to transit information packets by media servers, so we can say: media servers can be used as core component for VOIP. In this research work, assessing of media servers is done by simulators that they produce RTP's connections, in additions as an experimental components SEMS that it's a source of media server, is used for asserting the quality by doing packets information with SIP. We can observe that the more increase of connections and pass of the certain threshold, the less of quality. In addition, the other performance metrics such as error rate And packet lost are asserted. The identification of load saturation points and the efforts to eliminate disturbing factors during the increase of simultaneous call in this telephone system, can present quality-based approach to servers of these networks.

Multiple Transcoding Impact on Speech Quality in Ideal Network Conditions

This paper deals with the impact of transcod- ing on the speech quality. We have focused mainly on the transcoding between codecs without the negative in- fluence of the network parameters such as packet loss and delay. It has ensured objective and repeatable re- sults from our measurement. The measurement was performed on the Transcoding Measuring System devel- oped especially for this purpose. The system is based on the open source projects and is useful as a design tool for VoIP system administrators. The paper com- pares the most used codecs from the transcoding per- spective. The multiple transcoding between G711, GSM and G729 codecs were performed and the speech qual- ity of these calls was evaluated. The speech quality was measured by Perceptual Evaluation of Speech Quality method, which provides results in Mean Opinion Score used to describe the speech quality on a scale from 1 to 5. The obtained results indicate periodical speech quality degradation on every transcoding between two codecs.

Capture and Analysis of Malicious Traffic in VoIP Environments Using a Low Interaction Honeypot

The number of users of VoIP services is increasing every year. Consequently, VoIP systems get more attractive for attackers. This paper describes the implementation of a low interaction honeypot for monitoring illegal activities in VoIP environments. The honeypot operated during 92 days and collected 3502 events related to the SIP protocol. The analysis of the results allows understanding the modus operandi of the attacks targeted to VoIP infrastructures. These results may be used to improve defense mechanisms such as firewalls and intrusion detection systems.

Efficient ID-based authentication and key agreement protocols for the session initiation protocol

In a widely deployed VoIP system tens of thousands of clients compete for the SIP proxy server's authentication service. SIP protocol implementations have to meet certain QoS and security requirements. In this study new ID-based protocols are proposed for the SIP authentication and key agreement protocols. These protocols minimize the use of expensive pairing functions but still resist notable attacks. The security of the proposed protocols are analyzed and demonstrated with security proofs based on the BJM security model. Finally, the performance overhead of the proposed protocols are compared to ID-based SIP authentication and key agreement protocols given in the literature.

Exploiting VoIP softphone vulnerabilities to disable host computers: Attacks and mitigation

As increasing numbers of people run VoIP softphones on their laptops and smartphones, vulnerabilities in VoIP protocols and systems introduce new threats to the computer systems that run VoIP softphones. This paper investigates key threats that target VoIP hosts and techniques for mitigating the threats. In particular, this paper shows that crafted SIP traffic can disable a Windows XP host that runs a Vonage VoIP softphone or a Linphone by consuming almost all the free memory within minutes. While this “noisy attack” can be effectively mitigated by threshold-based filtering, the paper demonstrates that a “stealthy attack” can be launched to defeat threshold-based filtering and disable the host computer without ever ringing the softphone. To mitigate the stealthy attack, the paper describes a limited context aware filtering approach that leverages the context and SIP protocol information to ascertain the intentions of a SIP message on behalf of the client. Experiments demonstrate that limited context aware filtering can effectively defeat a stealthy attack while allowing legitimate VoIP traffic and calls to go through.

A game-theoretic analysis of preventing spam over Internet Telephony via audio CAPTCHA-based authentication

Spam over Internet Telephony SPIT is a potential source of disruption in Voice over IP VoIP systems. The use of anti-SPIT mechanisms, such as filters and audio CAPTCHA Completely Automated Public T...

A game-theoretic analysis of preventing spam over Internet Telephony via audio CAPTCHA-based authentication

Spam over Internet Telephony SPIT is a potential source of disruption in Voice over IP VoIP systems. The use of anti-SPIT mechanisms, such as filters and audio CAPTCHA Completely Automated Public Turing Test to Tell Computer and Humans Apart can prevent unsolicited calls and lead to less unwanted traffic. In this paper, we present a game-theoretic model, in which the game is played between SPIT senders and internet telephony users. The game includes call filters and audio CAPTCHA, so as to classify incoming calls as legitimate or malicious. We show how the resulting model can be used to decide upon the trade-offs present in this problem and help us predict the SPIT sender's behavior. We also highlight the advantages in terms of SPIT call reduction of merely introducing CAPTCHA, and provide experimental verification of our results.

Detection and Prevention Mechanism on Call Hijacking in VoIP System

Voice over Internet Protocol) enables user to make calls through internet. VoIP system is popular because of its rich features. VoIP uses Session Initiation Protocol for initialization, termination and tearing down of a session between two communicating entities. Due to its rich features VoIP has received wide acceptance and becoming one of the mainstreams in communication network, increased use of this includes scams and security concerns. Vulnerabilities in the SIP protocol enable hackers to inject control signals and hijack calls. Hence assuring the identities of the communicating entities is essential. Many authentication schemes were proposed for SIP from time to time. Strong authentication scheme can identify the potential threats. This paper proposes the authentication scheme between two end points in order to overcome the limitations of the existing authentication scheme.

Comparative analysis of traditional telephone and VoIP systems

Pakistan is in the period of upgrading communication infrastructure. Therefore many communication techniques are taking birth day by day and building the effective way of communication and one of many is IP Telephony system. In Pakistan the adoption of IP telephony is slow because of variety of reasons, but the main factor is the reliability of legacy phone systems. With a weak economy in Pakistan, the management team of organizations is happy to get as much mileage as possible out of their existing infrastructure, and if the legacy phones are working fine, IT team will need a really strong reason to change it to IP telephony. The purpose of this independent study is to drive the correct adoption of new IT & communications technologies in Pakistan to enable increased productivity and added value for businesses. This paper investigates the issues of service quality and cost lie in Pakistan and on the basis of this issues what steps we need to consider while deciding to implement VoIP. After implementation of VoIP in two different organizations and we then analysis their comparative results with legacy phone systems.

The design and implementation of SIP-B2BUA and its applications

For the special telephone communication needs of broadcast and TV, we discuss how to implement diversiform services in VoIP systems based on Session Initiation Protocol SIP. A new method based on SIP-B2BUA is proposed and then we implement it. First, a B2BUA component is designed and implemented. The component not only can do the basic functions of a B2BUA, but also has a smart dial-plan and an API of application modules. Second, it is discussed that the B2BUA call-processing ability extends for more diversiform services by the dial-plan smartly and application modules. The diversiform service includes callback service, Auto Call Distribution ACD and other call features of IP-PBX. Finally, we draw the general way out to implement diversiform services based on the B2BUA component.