VoIP System Research Articles

Predictive Modeling in a VoIP System

An important problem one needs to deal with in a Voice over IP system is server overload. One way for preventing such problems is to rely on prediction techniques for the incoming traffic, namely as to proactively scale the available resources. Anticipating the computational load induced on processors by incoming requests can be used to optimize load distribution and resource allocation. In this study, the authors look at how the user profiles, peak hours or call patterns are shaped for a real system and, in a second step, at constructing a model that is capable of predicting trends.

Survey on application-layer mechanisms for speech quality adaptation in VoIP

VoIP calls are sensitive to several impairments, such as delay and packet loss. One way to overcome these problems is by adaptively adjusting application-layer parameters to keep a minimum speech quality level. At the heart of self-adaptive systems lies a feedback loop , which consists of four key activities: monitoring, analysis, planning, and execution. Nevertheless, the existing adaptive approaches to QoS control of VoIP do not explicitly exhibit this feedback loop. Bringing it to surface can help developers in designing more robust and human-independent VoIP systems. This survey presents a comprehensive review of the current state-of-the-art research on speech quality adaptation of VoIP systems at the application layer and some research challenges on this subject.

Modified Digital Filtering Algorithm to Enhance Perceptual Evaluation of Speech Quality (PESQ) of VoIP

Speech quality of VoIP system is degraded due different network layer problems such as packet loss delay and jitter and external noise. This paper compares the quality of speech signal that is implemented on digital signal processor using G.729 audio data compression algorithm with the ITU-T G.711 PCM coder implemented using modified digital filtering algorithm. PESQ (ITU-T P.862, Perceptual Evaluation of Speech Quality) is used to evaluate the performance. The results indicate that our proposed architecture has better performance.

G.729 코덱의 패킷 손실 영향 모델을 이용한 비 침입적 음질 예측 기법

본 논문은 패킷 손실의 영향을 이용한 비 침입적 음질 평가 방법을 제안한다. 패킷 손실은 패킷 기반의 통신 시스템에서 음질을 저하시키는 주된 요소이며 그 영향은 코덱에 내장된 패킷 손실 은닉 알고리듬에 의해 결정된다. 패킷 손실 영향을 반영한 음질평가 시스템을 위해 VoIP 에서 협대역 코덱으로 사용되는 코덱 중 하나인 G.729를 선택하였으며, 음성 특징에 따른 패킷 손실 영향을 구분하기 위해서 G.729 코덱의 음성 파라미터를 이용한 한 음성 특성 분류기를 설계하였다. 이후, 각각의 패킷 특성에 따른 음질 저하의 정도를 수치화하기 위해 원 PESQ-LQ점수와 상관계수를 최대화하는 음질 저하 가중치를 반복적으로 구하였으며, 최종 음질 저하는 가중합으로 구하였다. 그 결과 제안한 모델과 PESQ-LQ의 상관계수는 칩입 모델 에서는 0.8950를, 비 침입 모델 에서는 0.8911의 결과를 나타내었다. This paper proposes a non-intrusive speech quality estimation method considering the effects of packet loss to perceptual quality. Packet loss is a major reason of quality degradation in a packet based speech communications network, whose effects are different according to the input speech characteristics or the performance of the embedded packet loss concealment (PLC) algorithm. For the quality estimation system that involves packet loss effects, we first observe the packet loss of G.729 codec which is one of narrowband codec in VoIP system. In order to quantify the lost packet affects, we design a classification algorithm only using speech parameters of G.729 decoder. Then, the degradation values of each class are iteratively selected that maximizes the correlation with the degradation PESQ-LQ scores, and total quality degradation is modeled by the weighted sum. From analyzing the correlation measures, we obtained correlation values of 0.8950 for the intrusive model and 0.8911 for the non-intrusive method.

Simulation Design and Performance Analysis for VoIP in Cognitive Radio Networks

Increasing popularity of VoIP systems has witnessed applications in emerging technologies like Cognitive Radio Networks (CRN). The stringent QoS requirements in VoIP coupled with complexities in CRN have initiated intensive research in the field of performance analysis and optimizations guided by simulation results. However, in the absence of any standard model of VoIP over CRN, accuracy and credibility of the simulation output are strongly dependent on proper design of the simulation model that must have a strong mathematical foundation. The objective of this paper is to build standard models for VoIP in CRN and successfully implement VoIP applications over CRN domain, which will serve as initial point for development with respect to all future simulation studies in VoIP over CRN category. Initially, models of VoIP in CRN are developed using OPNET Modeler 16.0.A following distributed architecture in single-channel and multi-channel scenarios and further in Visual C++ adhering to the principles of centralized architecture. The models are validated by comparison of simulation results obtained in both platforms. The underlying mathematical model behind the design is established and the critical factors pertaining to both VoIP and CRN domain are extensively analyzed.

A Method for Disguising Malformed SIP Messages to Evade SIP IDS

Malformed SIP attacks are threatening the security of VoIP system, such as IP Multimedia Subsystem, which uses SIP (Session Initiation Protocol) as its core protocol. Though IDSs (Intrusion Detection System) supporting malformed SIP detection had been produced, it was not clear to what extent they can detect disguised malformed SIP messages. This paper analyzes the condition of SIP IDS evasion and proposes a method for disguising malformed SIP messages. Based on the disguising method, a testing system is built for evaluation the capability of SIP IDS on evasion defending. The result of the experiments show that the proposed method can improve the evasion rate of malformed SIP messages considerably, which means the defending capability of SIP IDSs should be improved to prevent them from evasion.

A Bayesian Network-Based Knowledge Engineering Framework for IT Service Management

Service management is becoming more and more important within the area of IT management. How to efficiently manage and organize service in complicated IT service environments with frequent changes is a challenging issue. IT service and the related information from different sources are characterized as diverse, incomplete, heterogeneous, and geographically distributed. It is hard to consume these complicated services without knowledge assistant. To address this problem, a systematic way (with proposed toolsets and process) is proposed to tackle the challenges of acquisition, structuring, and refinement of structured knowledge. An integrated knowledge process is developed to guarantee the whole engineering procedure which utilizes Bayesian networks (BNs) as the knowledge model. This framework can be successfully applied on key tasks in service management, such as problem determination and change impact analysis, and a real example of Cisco VoIP system is introduced to show the usefulness of this method.

Voice Quality Evaluation of a Call Using Fuzzy Logic

Voice will remain a fundamental communication media that cut across people of all walks of life. It is there- fore important to make it very affordable. VoIP has been increasingly popular in recent times due to its affordability, how- ever, poor reliability and voice quality remain important factors that limit the widespread adoption of VoIP systems. Good voice quality is a key factor for users transiting from the Public Switched Telephone Network (PSTN) to VoIP networks. It has been shown that line echo is one of the key factors that deteriorate voice quality in VoIP systems. Several non-real-time algorithms have been developed in literature to estimate various aspects of voice quality in VoIP systems. But there is no real-time algorithm that estimates the echo content of a VoIP conversation, which could enable the operator take some cor- rective actions to improve the quality while the call is in progress. In this paper, the authors propose a real-time fuzzy algo- rithm to estimate the strength of the line echo component of the voice quality in VoIP networks. The results obtained shows that the algorithm is able to track and estimate echo content of a live VoIP traffic in real-time. This algorithm could be em- bedded in VoIP systems to enable operators monitors calls in real-time.

A lightweight payment scheme for real-time services based on SIP

In the session initiation protocol (SIP), payments have been proposed as a way for vendors to obtain profit from the services they provide. Payments in SIP have also been proposed for microbilling and even as a solution to SPAM in VoIP systems. Although several proposals exist for making payments in SIP, they present some limitations when we want to pay for access to real-time services: either they are not suitable for micropayments or they do not consider security in the payment information exchanged. As a response to these limitations, we propose a new SIP payment protocol, LP-SIP, that supports the payment according to different models like pay-per-time, session-based, etc. It also performs payments in SIP efficiently and takes into account the secure exchange of payment information, unlike other existing proposals. Thus, we provide a lightweight payment protocol that can be used for the payment of real-time services.

유무선 VoIP 취약점 분석과 해킹공격 및 보안 연구

Recently VoIP has provided voice(both wired and wireless from IP-based) as well as the transmission of multimedia information. VoIP used All-IP type, Gateway type, mVoIP etc. Wired and wireless VoIP has security vulnerabilities that VoIP call control signals, illegal eavesdropping, service misuse attacks, denial of service attack, as well as wireless vulnerabilities etc. from WiFi Zone. Therefore, the analysis of security vulnerabilities in wired and wireless VoIP and hacking incidents on security measures for research and study is needed. In this paper, VoIP (All-IP type, and for Gateway type) for system and network scanning, and, IP Phone to get the information and analysis of the vulnerability. All-IP type and Gateway type discovered about the vulnerability of VoIP hacking attacks (Denial of Service attacks, VoIP spam attacks) is carried out. And that is a real VoIP system installed and operated in the field of security measures through research and analysis is proposed.

User-Profile Framework against a Spit Attack on VoIP Systems

Spam over Internet Telephony is becoming a large problem of the Voice over IP Architectures. Voice over IP has become a valid alternative to the Public Switched Telephone Network and it has a new paradigm for providing telephony services at a lower cost and higher flexibility. In the meantime, with Voice over IP the privacy becomes so hard and the benefits introduced are as strong as the security problems, like Spam over Internet Telephony. People have more attention on the spam voice issue, because of the great danger of this threat. Thus, in this paper we want to propose and introduce a new protection model against Spam over Internet Telephony attack, through a user-profile framework. This framework allows to identify the optimal countermeasures to be taken against this threat, according to the user assessment, that receives a certain number of calls during an observation period. We define general profiles based on certain parameters, and with a user-profile matching we can identifies the best security path which must be applied.

An efficient complexity reduction algorithm for G.729 speech codec

Having been widely used in a VoIP system, the ITU-T G.729 standard is of high computational complexity, a heavy load for a real-time operation on a low-end smart phone or other embedded devices even with the G.729A standard. In this paper, there are three methods proposed for an efficient complexity reduction of G.729A, i.e., a combined local optimal pulse method, employed to search an algebraic codebook, a discontinuous pitch estimation, used to analyze the open-loop pitch, and the hybrid two-stage vector quantization (HTVQ), assigned to quantize the line spectral frequency (LSF) coefficients. It is validated by experimental results that the improved version of G.729A can be easily implemented on a smart phone (312 MHz CPU, 64 MB RAM) in real-time operation. In total 80.94% of computational load can be thus eliminated in comparison with the G.729A and a satisfactory speech quality is maintained as intended, an excellent efficiency demonstration by the methods proposed.

A Comprehensive Survey of Voice over IP Security Research

We present a comprehensive survey of Voice over IP security academic research, using a set of 245 publications forming a closed cross-citation set. We classify these papers according to an extended version of the VoIP Security Alliance (VoIPSA) Threat Taxonomy. Our goal is to provide a roadmap for researchers seeking to understand existing capabilities and to identify gaps in addressing the numerous threats and vulnerabilities present in VoIP systems. We discuss the implications of our findings with respect to vulnerabilities reported in a variety of VoIP products. We identify two specific problem areas (denial of service, and service abuse) as requiring significant more attention from the research community. We also find that the overwhelming majority of the surveyed work takes a black box view of VoIP systems that avoids examining their internal structure and implementation. Such an approach may miss the mark in terms of addressing the main sources of vulnerabilities, i.e., implementation bugs and misconfigurations. Finally, we argue for further work on understanding cross-protocol and cross-mechanism vulnerabilities (emergent properties), which are the byproduct of a highly complex system-of-systems and an indication of the issues in future large-scale systems.

Implementing Security on a Voice over Internet Protocol (VoIP) Network: A Practical Approach

The internet has been undergoing rapid and continuous growth over the past few years. The most significant development the telecommunication industry has witnessed is the evolution of Voice over IP technology. This technology uses the internet as a medium for transmitting voice during a telephone conversation between two or more parties. VoIP technology comes with numerous benefits like reduced call cost and wide variety of add-ons to both the service providers and the end users. Despite all these, the technology is subjected to security risk because of the open nature of the internet. Much emphasis has been placed on the quality of service of VoIP over the years and dark eyes have been turned on its security. This study therefore aims to identify the security threats and attack VoIP can be exposed to during transmission, produce a VoIP system using the open source Asterisks before securing the network using IPSec VPN

Binary CDMA 기반의 무선 VoIP 시스템의 구현 및 성능평가

Binary CDMA는 음성 영상 등의 멀티미디어 데이터를 초고속으로 우수한 전송품질로 통신할 수 있는 우리나라가 개발한 새로운 표준 무선통신 기술이다. 본 논문에서는 전화선의 설치나 매설 작업 없이 원하는 장소에 자유롭게 설치할 수 있는 Binary CDMA 기반의 무선 VoIP 시스템 개발을 위한 개발방법 및 구현에 관해 연구하였다. Binary-CDMA VoIP 시스템의 구조를 설계하고, 호제어를 위한 메세지와 자료구조의 정의한 후, 사용자간 통화를 위한 착발신 단말 간의 호제어 메세지 흐름을 설계하고 구현하였다. Binary CDMA is a new standard technology for wireless communication developed by our country that makes high speed communications and good quality of services. In this paper we researched the development methods and implementation of a wireless VoIP System based on the Binary-CDMA technology that makes it freely installed in any place without phone cables and laying works. We designed the structure of the Binary-CDMA Wireless VoIP System and defined messages and data structures for call control. And then we designed and implemented call control message flow between VoIP terminals for user's communication.

Binary CDMA 기반의 무선 VoIP 시스템의 개발방법 및 설계

Binary CDMA는 음성 영상 등의 멀티미디어 데이터를 초고속으로 우수한 전송품질로 통신할 수 있는 우리 나라가 개발한 새로운 표준 무선통신 기술이다. 본 논문에서는 전화선의 설치나 매설 작업 없이 원하는 장소에 자유롭게 설치할 수 있는 Binary CDMA 기반의 무선 VoIP 시스템 개발을 위한 개발방법 및 설계에 관해 연구하였다. Binary CDMA is a new standard technology for wireless communication developed by Korea that makes high speed communications and good quality of services. In this paper we researched on the development methods and design of a wireless VoIP system based on the Binary-CDMA technology that makes it freely installed in any place without phone cables and laying works.

SIG 18 Perspectives Vol. 1, No. 1, September 2011

SIG 18 Perspectives Vol. 1, No. 1, September 2011

A Novel Approach for Evaluating and Detecting Low Rate SIP Flooding Attack

over Internet protocol (VoIP) is continuously evolving and changing the face of business telephony. The Session Initiation Protocol (SIP) is a widely used standard in VoIP communications to setup and tear down phone calls. Low rate Denial-of Service (DoS) attack recently emerged as the greatest threat to enterprise VoIP systems. Such attacks are difficult to detect and capable of discovering vulnerabilities in protocols with low rate traffic. In this paper we aim to p rovide a novel low rate SIP flooding detection scheme using area under curve of monitored dynamic SIP traffic with classification of SIP flooding attacks and its impact on SIP server under low rate DoS attack. Compared to the other detection system our technique achieves advantages of accuracy, fast, light weight, and flexibility to deal with DDoS attack detection. Experimental results show the effectiveness of the scheme.

ITDTS: A SIP-based Telephone System for Train Dispatching

Train Dispatching Telephone System(TDTS) is a telephone system works for communication among train, railway station and railway dispatching center. The traditional TDTS works on 400MHz mode for voice analog, making it difficult to realize end-to-end communication. With the application of wireless access networks in railway, VoIP-based TDTS is tending to be a reality. Different from ordinary VoIP system, clients in TDTS have multi-level precedence. Client with high level precedence can disconnect or join in an existing session consisting of low level clients. For example, when emergency happens, the dispatcher, which has the highest precedence, can deliver the dispatching command to a busy station or a busy train driver through force-insert or force-disconnect. The call flow of force-insert and force-disconnect are also different from the call flow of the ordinary call. Therefore, this paper firstly gives a formalized system model to describe the relationship among different roles, different precedence and multiple call patterns. Based on the system model, this paper focuses on the analysis and design of the call flows of precedence based force-disconnect call and force-insert call. Although the three kinds of roles(train, station, and dispatcher) in TDTS have different precedence, they have almost the same service logic, namely, signaling processing, media processing. So we design a MVC(Model-View-Control) based GUI framework for the roles. Each kind of roles can be easily developed and maintained under such GUI framework. Utilizing session initiation protocol (SIP), we design and develop the whole system iTDTS. The performance of iTDTS is finally evaluated in a real railway line over IEEE 802.16. Our efforts show that iTDTS works well over IEEE 802.16 networks.

A VoIP system for mobility voice security support using the VPN

ABSTRACTVoice over internet protocol (VoIP) provided voice service using Internet. It receives footlights when it escapes an initial curiosity. Interest to VoIP increased, because it can transfer existing phone service and deliver voice data through Internet technology. Communication service providers introduce Next Generation Network and Broadband convergence Network with VoIP as an added service to the consumer. In addition, session initiation protocol (SIP) service can be applied even outside the Internet phone service. However, it is needed to ensure secrecy of VoIP call in a special situation. It is relatively difficult to eavesdrop in the commonly used Public Switched Telephone Network (PSTN) as it is connected with 1:1 circuit. In this paper, we propose a new model of Internet telephone for eavesdrop prevention enabling VoIP (using SIP protocol) to use the virtual private network (VPN) protocol and establish the probability of practical use comparing it with Internet telephone. Copyright © 2010 John Wiley & Sons, Ltd.