Securing VoIP Services in Multi-Hop Wireless Mesh Networks

Voice over Internet Protocol (VoIP) service has been a very popular and important application over the Internet. Recently, wireless mesh networks (WMNs) has been considered as a good solution for VoIP services due to low cost and convenience, easy deployment and larger area coverage. However, many relevant security issues have been raised for integrating IP Telephony into the existent applications and system infrastructure. In this paper, we propose a trust degree-based dynamic model for securing the VoIP over WMNs, in which each node maintains trust degree for other nodes and applies the trust degree to evaluate their trustworthiness. We also validate the performance of our model with simulation results

Recently, plenty of voice quality measuring devices of VoIP(Voice over Internet Protocol) service is developed to support high quality VoIP service for communication. However, the existing voice quality measuring devices select the unique measurement method, which uses PESQ(Perceptual Evaluation of Speech Quality) or R-factor only. However, the each voice quality measurement method differs from its usage and assumed range, which leads to the slightly different results at the same testing area for voice services. At the actual VoIP service business and related regulatory organizations, they request to use the above two types of algorithm at the same time for measuring voice quality. However, the development expenses are very high to use the above two types of algorithm at the same time for measuring voice quality due to additional resources and subsidiary devices. VoIP business and regulatory agencies also request the hardware specification of the voice quality measurement device is similar to that of VoIP service user terminal, considering VoIP service supply at the wireless mobile internet such as WiBro(Wireless Broadband). In this paper, we present the voice quality measurement method by using of the above two algorithms at the one physical measurement device at the same time for more accurate and reliable VoIP voice quality measurement. In addition, we introduce the development method of the voice quality measurement device with relatively low cost.

Systems employing the Internet and Voice over Internet Protocol (VoIP) are alternative to the legacy landline telephony system. The services offered by these systems allow users to communicate with their family members, friends, banks and business partners whenever they are online. Particularly, the VoIP service is popular with the Internet users because charges are bound into usually fixed access costs making the price of the long distance calls themselves appear economical and even free. However, VoIP services have several disadvantages. One of the disadvantages of VoIP is, it is exposed to Internet security vulnerabilities, threats and attacks. The attacks come in many forms. One of the methods to attack the VoIP service, is by sending a Spam over Internet Telephony (SPIT). SPIT is normally executed by malicious parties initiating automated, unsolicited, and unwanted communications that used VoIP or video conferencing services, just like an email SPAM. Solutions to mitigate SPIT are still lacking. To defense a VoIP system from SPIT, we have come out with the idea of Automated Intrusion Detection and Prevention System over SPIT (AIDPoS) that can detect and prevent SPIT from external parties to enter the VoIP network. AIDPoS is a combination system of Intrusion Detection System (IDS) and Intrusion Prevention System (IPS) on Voice over IP (VoIP) server. This is an experimental research to find a solution to combat SPIT if it happens in the future. Our result shows that an automated Intrusion detection and Prevention system over SPIT attack is feasible. Our work is on an open source VoIP network. Other vulnerabilities and threats are important, but beyond our research scope and is not covered in this paper.

The significant increase in Voice over IP (VoIP) service providers over the last few years has prompted an increase in VoIP services and currently poses additional challenges when deployed on wireless mesh network. Packet losses and delay due to interference in a wireless mesh network degrade significantly the VoIP call quality. In this work, we propose an approach inspired of Traffic Engineering Concept for wireless mesh network to provide differentiated services and therefore a better Quality of Service. We establish general criteria for these algorithms: routing using of calls statistics, minimizing network costs, and load balancing. Finally, we propose a solution that encompasses the different criteria presented in the paper.

This paper has evaluated the system performance for the voice over Internet protocol (VoIP) services supported by the IEEE 802.16 systems in terms of the packet loss and delay, and it has also calculated the mean opinion score (MOS) in order to estimate the user's satisfaction for the VoIP services. By the simulation results, the MOS of the VoIP service supported by the unsolicited grant service (UGS) without auto repeat request (ARQ) has been decreased by 1.5 when the channel quality is poor. The reason of the MOS reduction is the increase of the packet loss due to the channel error. For this reason, this paper has considered the ARQ in order to compensate the packet loss when the channel quality is poor. The simulation results have shown that the extended real-time polling service (ertPS) and rtPS with ARQ can compensate the packet loss even if the channel quality is poor, however they can cause the reduction of the VoIP capacity because of the resource used for the retransmission. From the simulation results, the ertPS could save the resource by about 25 % compared to the rtPS with guaranteeing a quality of service (QoS).

Voice over Internet Protocol (VoIP) over wireless mesh networks (WMNs) is becoming more and more popular due to its significant cost advantage, higher flexibility and wide coverage. One of the most basic requirements of VoIP is that it must provide privacy and security protection for VoIP clients. Unfortunately, little effort is spent on this area. In this paper, we focus on the privacy security of session initiation protocol (SIP) and propose a privacy-aware SIP called PA-SIP to enhance the VoIP privacy security. PA-SIP extends the SIP and applies a subjective logic based reputation mechanism to detect and isolate the insider malicious nodes. Finally, we analyze the PA-SIP in terms of security and performance.

One of the most common ways, where anyone can communicate with each other very easily, and instantly, is, of course, the voice exchange. Public Switched Telephone Networks (PSTN), and Voice over Internet Protocol (VoIP) are common ways of transferring voice communications by using telephone network. These two ways have their own strengths and weaknesses. The two major positive points of the PSTN are reliability, and stability. One of the drawbacks of the PSTN is that it has complex connectivity. The VoIP strengths are cheaper in costs compared to PSTN, less usage of cable, user-friendly, simple connectivity, and variety of services. On the other hand, the biggest challenge of VoIP is frequent disconnects of voice communication. In this paper, a survey is being performed on a selected group of people to understand the quality differences between PSTN, and VoIP. This paper shows that the Quality of Service (QoS) of today's VoIP is not good enough but users still continue to use it. However, this survey is also trying to find ways on how to improve existing QoS of VoIP. Plus, it points out what type of users prefers to utilize by using PSTN or VoIP.

Voice Over Internet Protocol (VoIP) is an important technology in a network Communication system. In the Communication System, security is an important factor. Nowadays, session initial protocol is mostly used in VoIP Services. Session Initiation Protocol is used for many security risks, as well as for flooding and developed in an inclusive environment. Flooding is an attack called Denial of Services attack, a cyber-attack in which an attacker disrupts the user traffic by flooding data on traffic. Different technologies and methods to warning the users to prevent from attacks are discussed in this paper. This paper also describes the vulnerability and tests related to VoIP that are used in VoIP Security tools to find the weaknesses that create threats and issues in the security of Voice over Internet Protocol.

LTE (Long Term Evolution), a 4G mobile communication standard has a support for higher data rates and lower latency. Since LTE is completely an Internet Protocol (IP) based technology, Network operators of LTE can handle voice via a “fall-back” technique by using previously existed circuit switched networks. To avoid this, Voice over IP (VoIP) services need to be used to deliver the voice over IP networks, such as Internet. Voice Codec is one of the critical components in VoIP used to convert analog signals to compressed digital packets and vice-versa that can be sent over the internet. But there are different kinds of codecs exist that optimize the voice stream based on bandwidth and application requirement. Unfortunately, there is no standard codec that is acceptable for all circumstances, some codecs provides better quality, some other uses lesser bandwidth. All these features will effect on the ultimate selection of codec by an organization. This paper focuses on the quality of service (QOS) analysis of VoIP traffic over LTE by configuring different voice codecs such as G.711, G.722.2, G.723.1 and G.729. To analyze the performance of these codecs ns-3 simulator is being used since its core has a support for LTE models. The comparative study of above selected VoIP codecs over LTE network in terms of Packet Loss, End to End delay, Jitter, and Throughput in normal and error prone scenarios has been done. From the analyzed results and the performance evaluated, the selection of codec will become easy to provide better VoIP services for customer satisfaction.

Design and realization of ad-hoc VoIP with embedded p-SIP server

Voice over IP (VoIP) over a wireless mesh network (WMN) is a killer application candidate of the next generation services over wireless and mobile network infrastructure. Yet, the routing and transport efficiencies in consideration of specific attributes of WMNs have been little emphasized. Distinctively from conventional networks, the major traffic pattern of WMNs is anycast (one-to-one-of-many). To fully utilize the service aspects of WMNs, load balancing and scalability should simultaneously be supported. For VoIP, one challenge is to maintain the perceived quality depending on both loss and delay. In this paper, we assess autonomous load balancing anycast routing regarding VoIP supportability. The protocol is based on physics analogy and a distributed algorithm. We compare the protocol with shortest-path routing and back-pressure routing via simulations. Simulation results exhibit that autonomous load balancing of the protocol efficiently supports voice calls to guarantee the certain level of quality.

The usage of voice over IP (VoIP) traffic in IEEE 802.11 wireless networks is expected to increase in the near future due to widely deployed 802.11 wireless networks and VoIP services on fixed lines. However, the quality of service (QoS) of VoIP traffic in wireless networks is still unsatisfactory. In this thesis, I identify several sources for the QoS problems of VoIP traffic in IEEE 802.11 wireless networks and propose solutions for these problems. The QoS problems discussed can be divided into three categories, namely, user mobility, VoIP capacity, and call admission control. User mobility causes network disruptions during handoffs. In order to reduce the handoff time between Access Points (APs), I propose a new handoff algorithm, Selective Scanning and Caching, which finds available APs by scanning a minimum number of channels and furthermore allows clients to perform handoffs without scanning, by caching AP information. I also describe a new architecture for the client and server side for seamless IP layer handoffs, which are caused when mobile clients change the subnet due to layer 2 handoffs. I also present two methods to improve VoIP capacity for 802.11 networks, Adaptive Priority Control (APC) and Dynamic Point Coordination Function (DPCF). APC is a new packet scheduling algorithm at the AP and improves the capacity by balancing the uplink and downlink delay of VoIP traffic, and DPCF uses a polling based protocol and minimizes the bandwidth wasted from unnecessary polling, using a dynamic polling list. Additionally, I estimated the capacity for VoIP traffic in IEEE 802.11 wireless networks via theoretical analysis, simulations, and experiments in a wireless test-bed and show how to avoid mistakes in the measurements and comparisons. Finally, to protect the QoS for existing VoIP calls while maximizing the channel utilization, I propose a novel admission control algorithm called QP-CAT (Queue size Prediction using Computation of Additional Transmission), which accurately predicts the impact of new voice calls by virtually transmitting virtual new VoIP traffic.

Wireless Mesh Networks (WMNs) have gained prominence in modern communication technology due to their flexibility and ease of deployment, which are advantageous in scenarios like disaster management and rescue operations. However, existing methods for enhancing the performance of WMNs, such as increasing the number of gateways, are costly, introduce interference, and complicate deployment. Moreover, current routing protocols often suffer from suboptimal packet delivery due to inadequate traffic flow management and packet loss. This research addresses these gaps by proposing a novel optimization model that integrates Artificial Bee Colony (ABC) and Particle Swarm Optimization (PSO) techniques to enhance packet delivery ratio in WMNs using Voice over Internet Protocol (VoIP). Unlike traditional approaches that overlook efficient traffic management, our proposed model focuses on optimizing packet transmission by selecting efficient routes and minimizing packet loss. The novelty of this solution lies in its hybrid use of ABC and PSO for dynamic node and route selection, which significantly improves network performance, reduces control overhead, and minimizes packet loss. Experimental results demonstrate that the proposed model outperforms existing protocols, making it a promising approach for enhancing network reliability and efficiency in WMNs.

Accurate modeling of VoIP traffic QoS parameters in current and future networks with multifractal and Markov models

On the development of the various IP (Internet protocol)-based services, it is expected that Internet telephony service will gradually replace the traditional PSTN (public switched telephone network)-based telephony service. But there are many issues to be resolved in order to spread widely the Internet telephony services. One of them is to support SMS (short message service) through the heterogeneous VoIP (voice over IP) environment. Even though the SMS has been used as one of the most famous services in the cellular network, it should be also provided in the ubiquitous environment that supports any services to anyone at anytime, anywhere with any devices. In this paper, we suggest a new mechanism for supporting the SMS on the heterogeneous VoIP environment; specifically H.323 and SIP (session initiation protocol) coexist in the Internet. This mechanism is expected to help in providing better quality of services to the ISPs and Internet users on the emerging VoIP systems and services markets.