A resource allocation scheme for VoIP traffic under high speed train scenario

Voice over Internet protocol (VoIP) is one of the most important applications for IEEE 802.11 wireless local area networks (WLANs). For network planners who deploy VoIP over WLANs, one of the important issues is the VoIP capacity. Directly implementing VoIP over infrastructure WLANs will create the bottleneck problem at the access point (AP). In this paper, we propose the use of the service differentiation provided by the new IEEE 802.11e standard to solve the bottleneck problem and improve voice capacity. In particular, we propose the allocation of a higher priority access category (AC) to the AP while allocating lower priority AC to mobile stations. We develop a simple Markov chain model, which considers the important enhanced distributed channel access (EDCA) parameters and the channel errors under saturation and nonsaturation conditions. Based on the developed analytical model, we analyze the performance of VoIP over EDCA. By appropriately selecting the EDCA parameters, we are able to differentiate the services for the downlink and uplink. The experimental results are very promising. With the adjustment of only one EDCA parameter, we improve the VoIP capacity by 20%-30%.

In this paper, we propose a throughput optimization scheme for relay protocols based on Adaptive Modulation and Coding (AMC). The performance of relay protocols with the AMC scheme and show that relay protocols with the AMC scheme are capable of providing better average throughput at a lower Signal to Noise Ratio (SNR) level as compared to the conventional scheme with no AMC. We perform 3GPP Long Term Evolution-Advanced (LTE-A) parameters based simulations to prove the performance comparison of adaptive Modulation and Coding Scheme (MCS) relay protocols with non-adaptive MCS relay protocols. The simulation results of the proposed system with adaptive MCS prove that among the Amplify-and-Forward (AF), Decode-and-Forward (DF) and De-Modulate-and-Forward (DMF), the DMF protocol performs best, at a lower SNR value and higher average throughput.

Information technology is developing rapidly, especially in the field of computer science, telecommunications and information systems. In the early days of telecommunications, voice networks could not be combined with data networks. However, more recently it became possible to integrate data, audio and video services onto a single network, known as a multimedia network. The capabilities of the Internet are also rapidly increasing, and now, in addition to data packets (email, web browsing) the Internet transfers audio and video packets. Voice over Internet Protocol (VoIP) is a multimedia service. This technology can be used to transmit audio, video, and data packets. Consequently, it is becoming the most preferred communications technology. VoIP will eventually replace the use of traditional telephony – Public Switched Telephone Networks (PSTN), although the switchover process is challenging. A chief concern it that, while PSTN transmits audio packets over a closed network, VoIP sends audio packets using an open network – the Internet. In PSTN, eavesdroppers must have direct access to physical network to obtain information from communications. Whereas with VoIP, eavesdroppers can monitor data packets from they are connected to the Internet. Hence, security and privacy are important considerations when switching to VoIP systems. One solution is anonymous systems, which can be used to implement security and privacy protocols. However, this typically reduces the Quality of Service (QoS) of the VoIP. This empirical research study examines VoIP performance in an anonymous network – The Onion Routing (Tor). Two scenarios are implemented using the real Tor network to investigate three QoS metrics for VoIP: latency, jitter and packet loss. As recommended, latency in VoIP should be less than 400 ms, jitter should be less than 50 ms and packet loss should not be more than 5%. In addition, the research calculates the probability of attackers in the two scenarios implemented and evaluate Tor network forecasting. Experiments were conducted in reference to two scenarios. The first scenario is VoIP calls routed through a Tor network with three Tor relays (default Tor). And the second scenario is VoIP calls routed through a Tor network with two Tor relays. Experiments were performed in three periods; December 2012, July 2013, and October 2013. During each experimental time period, a hundred calls were captured for each scenario. Experimental results show that the QoS of VoIP over the two Tor relays was better than the VoIP over the three Tor relays. However, a probability of attackers calculation found that the VoIP with three Tor relays returned better anonymity than that with two Tor relays. The best results were returned over the experimental period in July 2013, when the acceptable calls using two Tor relays reached 64 calls at 5% packet loss. Meanwhile, the worst experimental results were returned in October 2013, when acceptable calls numbered 11 using three Tor relays at 1% packet loss. At the end of 2013, the actual data for relay numbers and bandwidth approached forecast result with time series analysis. Meanwhile, the numbers of Tor users differed from the Tor users forecast. In August 2013, Tor users increased dramatically; this is attributed to the fact that BotNet attack was using the Tor network to attack their target leading to a fivefold increase.

A cross-layer algorithm for geographic routing in wireless sensor networks (WSNs) is proposed, which is robust to dead-ends and resilient to topological variations due to network dynamics. The solution combines ideas of network tessellation (clusterization) with greedy forwarding, without suffering from the problems afflicting landmark-based alternatives. The clusterization algorithm is based on a discovered graph-spectral property and relies on connectivity information only. Cluster sizes can be varied, allowing for different trade-offs between packet delivery success ratio (PDSR) and average packet delivery latency (APDL) to be reached. Simulation results show that the technique can substantially improve the PDSR in networks where large concave holes (dead-ends) are present, with no or little impact on APDL.

The lecture will present a test of the VoIP (Voice over Internet Protocol) network at the University of Applied Sciences (FHTW) Berlin based on Cisco VoIP phones model 7970G as an example in test of security policy of VoIP networks to find out the security gaps and to support improving security of the FHTW's VoIP campus network. Configuration options of the VoIP phone used at the University of Applied Sciences Berlin will be presented. The finding, installing and configuring products for testing the security of VoIP network and VoIP phones have to be dicussed and the security test of Cisco VoIP phones presented offer ways for make more secure configuration and steps to improve the overall security of VoIP networks. The presented results are backed by the long term ERASMUS TN co-operation between the University of Applied Sciences Berlin and the University of Rousse.

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.

Long Term Evolution (LTE) network offers a high throughput with low latency that makes it the best choice for Evolved Multimedia Broadcast Multicast Service (E-MBMS). E-MBMS transmits data to all users simultaneously using the same channel and same Modulation and Coding Scheme (MCS) level. Each user experiences different channel condition, so an MCS level should be selected properly to satisfy all users. Typically, MCS level is selected on the basis of users' worst channel condition which results in wasting network resources. Since the E-MBMS performance directly depends on the selected MCS level, an efficient Adaptive Modulation and Coding (AMC) scheme is essentially needed. This paper introduces a new mechanism that uses the Standard Deviation (SD) of MCS level of E-MBMS user group operated on single cell mode to efficiently select its optimal MCS level. Simulation results show that the proposed mechanism increase the spectrum efficiency, throughput, and lower Packet Loss Ratio compared to the Worst User Channel Condition mechanism and Fixed Predefined MCS mechanism.

In order to provide the same or better service quality in the Internet than traditional circuit-switched telephone network, a number of issues have to be solved which have hampered it in the Internet. Therefore, in this paper the VoIP (Voice over Internet Protocol) network design problems are modeled as nonlinear non convex combinatorial integer mathematical programming problems. The optimization problems have great practice value for the network service providers. This problem is a constrained multicast QoS routing and is proved to be a NP complete problem. The total link capacity augmented cost should be minimized when we design a new VoIP network or when the original network could not serve all of the traffic demands. It is also known as a complicated problem. Introduction Originally, the Internet was designed to provide best-effort services for the data generated by computers. The timeliness of data communication is generally delay tolerant. Quality-of-Service (QoS) constraints are not as important as routing flexibility and connectivity. Hence, using IP to transport voice data is contradictory to the basic requirement of the voice service: a timely delivery of voice samples. Although IP was not initially designed to provide services for real-time traffic, recent technical progress has made IP have the capabilities to provide real-time services in near future. In order to provide the same or better service quality in the Internet than traditional circuit-switched telephone network, we must deal with a number of issues that have hampered it in the Internet. Voice service requirements could be discussed from two perspectives: (1) application requirements such as end-to-end delay, jitter, packet loss and overdue probability; (2) user’s perspective such as reliability, availability, and supplementary services [1][2]. Telephony service providers must guarantee the quality of service they provide e.g. maximum one way delay does not exceed 150ms. Internet telephony requires a range of protocols, ranging from those needed for transporting real-time data across to the network e.g. Real-time Transport protocol (RTP) [3], to Quality-of-Service aware routing (QoS routing), signaling protocol, resource reservation, internetworking between IP networks and PSTN, QoS-aware network management and billing protocols. ITU-T defined H.323 to provide multimedia communication in packet networks. From the perspective of network service providers, they want to optimize the network performance such as minimizing the total bandwidth consumption, maximizing throughput or total revenue [4][5] subject to user and application constraints. In this paper, we want to develop the mathematical model for VoIP network. We minimize the total bandwidth consumption under users’ QoS requirements, the network topology and the network capacity. Performance Optimization Model It is generally accepted that Internet telephony and traditional circuit-switched telephony will coexist for quite some time. The VoIP architecture must deal with interworking between IP networks and PSTN, so we need gateways between the two worlds. There are four possible models of VoIP [6]. They are PC-to-PC, Gateway-to-Gateway, PC-to-Gateway, and Gateway-to-PC models. The architecture of VoIP is shown in Fig.1. The first model of VoIP is PC-to-PC architecture, which based 2nd International Conference on Electrical, Computer Engineering and Electronics (ICECEE 2015) © 2015. The authors Published by Atlantis Press 1144 on the assumption that two or more users have access to multimedia computers that are connected to the Internet. Fig.1 VoIP Architecture The VoIP system is modeled as a graph, where the hosts, VoIP gateways, and switches are represented by nodes and communication link sets are represented by links. Let N = {1, 2,..., n} be the set of nodes and L be the set of links in the graph (network). Let G be the set of all user groups. An user group g is a voice communication session requesting for transmission in the network. An user group g may be an unicast from the source to a destination or a multicast from the source to multiple destinations. For each user group g, the traffic is transmitted exactly over one tree. Dg represents the set of destinations of user group g. g λ is the required bandwidth of the voice transmission for each user group g. Below is a verbal description of the VoIP system design problem we considered. Given : Network topology Capacities of network links Equivalent bandwidth of each multicast/unicast group Time threshold and tolerable overdue probability for each multicast/unicast group To determine : The minimum overall bandwidth consumption Routing tree for each multicast/unicast group Overdue probability of each multicast/unicast group Objective : To minimize the total bandwidth consumption Subject to : End-to-end QoS (overdue probability) constraint Tree constraint Multi-commodity flow constraint Capacity constraint Integrality constraint Hop constraint Table 1 and 2 are the notations we use in this paper . Internet VoIP Gateway VoIPGatewa Central office switch or local hub Central office switch or local hub Multimedia PC IP Router Multimedia PC IP Router Gatekeepe r

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).

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.

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.

Emerging markets are increasingly competitive under the heading of convergence and substitution, including device convergence, fixed-mobile convergence, fixed-mobile substitution, service convergence, VoIP (Voice over Internet Protocol) telephony substituting for circuit switched voice telephony, bundled offers of mobile, fixed, broadband and TV and, finally, truly unified communications and digital media services that are delivered irrespective of anytime, anywhere, and anyhow. With the increasing use of wireless networking, convergence technology combined with user mobility is expected to introduce a significant change in how and when people communicate, using a variety of new applications on ubiquitous wireless devices. This chapter aims to discuss VoIP over enterprise fixed mobile converged networks (FMCN) and points out some open issues. We first review VoIP basics including transport and signalling protocols, speech coding, and VoIP quality assessment model, followed by introducing enterprise FMCN architecture and service characteristics. Then we review the performance of VoIP over enterprise wireless networks and discuss voice over enterprise unified communications. Finally, several future directions including VoIP over femtocell, location/presence-enriched enterprise VoIP, and admission control in enterprise wireless network are briefly listed.

This chapter starts by presenting the Stanford University Interim (SUI) and modified Friis propagation models. Although the SUI model is being recommended for WiMAX, the comparison between the model and experimental results show that, in our environment, at 3.5 GHz, the modified Friis model with g = 3 fits better the measurement values. From the analyses of the signal-to-noise-plus-interference ratio, SNIR, interference-to-noise ratio and reuse pattern, it is found that both noise and interference present a strong limitation to the performance of fixed WiMAX, mainly for higher order modulation and coding schemes (MCSs). In general terms, the use of sectorization in fixed WiMAX enables to reduce the reuse pattern while considering sub-channelisation allows for improvement on the coverage. The reduction of the reuse pattern directly corresponds to an increase in the system capacity but the improvement in the coverage range (through sub-channelisation) can also allow for an improvement in UL system capacity, as adaptive MCS are used. Two different approaches are considered for graphical cellular planning, and the district of Covilha is considered as a case study. On the one hand, one considered a GIS based WiMAX planning tool conceived by considering coverage issues, frequency reuse, and the impact of the different classes of service. On the other, Winprop™ is used as it distinguishes among different MCS in the graphical presentation of the results. Both tools consider the information coming from the digital terrain profile. The GIS functionalities allow for appropriately adjusting azimuth and tilt of antennas. This cellular planning exercises confirm the results of theoretical analysis, where different crowns are achieved for the coverage with each MCS (corresponding to a given range of values for SNIR), for the maximum physical throughput, and for the “best server“ cells. The frequency radio resources should be considered as the most valuable resource during the planning of wireless broadband access networks. As a rule, spectral efficiency needs to be optimized by using several advanced techniques, corresponding to an optimization from the cost-benefit point of view.

Voice over Internet protocol (VoIP) has become a popular alternative to traditional public-switched telephone network (PSTN) networks that provides advantages of low cost and flexible advanced ldquodigitalrdquo features. The flexibility of the VoIP system and the convergence of voice and data networks brings with it additional security risks. These are in addition to the common security concerns faced by the underlying IP data network facilities that a VoIP system relies on. The result being that the VoIP network further complicates the security assurance mission faced by enterprises employing this technology. It is time to document various security issues that a VoIP infrastructure may face and analyze the challenges and solutions that may guide future research and development efforts. In this paper, we examine and investigate the concerns and requirements of VoIP security. After a thorough review of security issues and defense mechanisms, we focus on attacks and countermeasures unique to VoIP systems that are essential for current and future VoIP implantations. Then, we analyze two popular industry best practices for securing VoIP networks and conclude this paper with further discussion on future research directions. This paper aims to direct future research efforts and to offer helpful guidelines for practitioners.

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