Voice Over IP Performance Research Articles

Towards a Smart Environment: Optimization of WLAN Technologies to Enable Concurrent Smart Services

In this research paper, the spatial distributions of five different services-Voice over Internet Protocol (VoIP), Video Conferencing (VC), Hypertext Transfer Protocol (HTTP), and Electronic Mail-are investigated using three different approaches: circular, random, and uniform approaches. The amount of each service varies from one to another. In certain distinct settings, which are collectively referred to as mixed applications, a variety of services are activated and configured at predetermined percentages. These services run simultaneously. Furthermore, this paper has established a new algorithm to assess both the real-time and best-effort services of the various IEEE 802.11 technologies, describing the best networking architecture as either a Basic Service Set (BSS), an Extended Service Set (ESS), or an Independent Basic Service Set (IBSS). Due to this fact, the purpose of our research is to provide the user or client with an analysis that suggests a suitable technology and network configuration without wasting resources on unnecessary technologies or requiring a complete re-setup. In this context, this paper presents a network prioritization framework for enabling smart environments to determine an appropriate WLAN standard or a combination of standards that best supports a specific set of smart network applications in a specified environment. A network QoS modeling technique for smart services has been derived for assessing best-effort HTTP and FTP, and the real-time performance of VoIP and VC services enabled via IEEE 802.11 protocols in order to discover more optimal network architecture. A number of IEEE 802.11 technologies have been ranked by using the proposed network optimization technique with separate case studies for the circular, random, and uniform geographical distributions of smart services. The performance of the proposed framework is validated using a realistic smart environment simulation setting, considering both real-time and best-effort services as case studies with a range of metrics related to smart environments.

Investigation of Techniques for VoIP Frame Aggregation Over A-MPDU 802.11n

The widespread and desirable features of IP and IEEE 802.11 networks have made these technologies a suitable medium for carrying voice over IP (VoIP). However, a bandwidth (BW) exploitation obstacle emerges when 802.11 networks are used to carry VoIP traffic. This BW exploitation obstacle is caused by the large 80-byte preamble size of the VoIP packet and a waiting time of 765 μs for each layer 2 VoIP frame. As a solution, IEEE 802.11n was consequently designed with a built-in layer 2 frame aggregation feature, but the adverse impact on the VoIP performance still needed to be addressed. Subsequent VoIP research has proposed numerous enhancements to the built-in IEEE 802.11n aggregation techniques. In this study, we analyse and propose enhancements to the BW exploitation and VoIP performance over IEEE 802.11n. We also recommend new techniques to enhance VoIP performance via IEEE 802.11n frame aggregation implementations. Furthermore, we contribute guidelines for creating a robust, suitable, and reliable aggregation technique for VoIP over the 802.11n standard.

ENHANCEMENT OF VoIP PERFROMANCE IN MANET USING FUZZY LOGIC

Voiceover Internet Protocol (VoIP) application is a vital technology that is quickly growing in the Mobile Ad hoc NETwork (MANET). Packet loss is a factor that can significantly affect the Quality of Service (QoS) for VoIP performance. Due to the dynamic nature of MANET, it is a challenging task to maintain the desired packet loss rate. This paper aims to enhance the performance of VoIP in the MANET using a fuzzy logic model. The input for the model is VoIP packet loss and the outputs are the optimal parameters of MANET (node number, pause time, maximum speed, and maximum connection).Network Simulator (NS2) was used to perform all simulations. MATLAB was used to implement the proposed fuzzy model. Moreover, the performance of the model was evaluated using NS2, and the results show that our proposed fuzzy model offers a significant enhancement in terms of the VoIP packet loss rate (P.LR).

Performance Investigation of VoIP Over Mobile WiMAX Networks through OPNET Simulation

Worldwide Interoperability for Microwave Access (WiMAX) is regarded as a promising technology that can provide wireless communication because of its advantages which include, high-speed data rates, high coverage and low cost of development and maintenance. WiMAX also supports the performance of Voice over Internet Protocol (VoIP), which is expected to replace conventional circuit switched voice services. VoIP requires to accurately design of QoS configurations over WiMAX networks. This paper focuses on studying and analyzing the performance of VoIP over WiMAX mobile networks. WiMAX network and VoIP technology such as mobility, WiMAX service classes, number of nodes and VoIP codecs are studied and analyzed. WiMAX network is simulated in a different manner using the simulation program known as OPNET Modeler. Simulation results established that the service layer Unsolicited Grand Service (UGS) is more appropriate for VoIP service because it has the best standard and performance. It was also observed that the least delay and highest value of customer satisfaction rate of services is demonstrated by the G.723.1 best coding. It also maintains the minimum consumption of capacity.

Implementing a secure VoIP communication over SIP-based networks

Recent years the Session Initiation Protocol (SIP) is commonly used in establishing Voice over IP (VoIP) calls and has become the centerpiece for most VoIP architecture. As wireless and mobile all-IP networks become prosperous, free VoIP applications are utilized in all places. Consequently, the security VoIP is a crucial requirements for its adoption. Many authentication and key agreement schemes are proposed to protect the SIP messages, however, lacking concrete implementations. The performance of VoIP is critical for users’ impressions. In view of this, this paper studies the performance impact of using key agreements, elliptic curve Diffie–Hellman and elliptic curve Menezes–Qu–Vanstone, for making a SIP-based VoIP call. We evaluate the key agreement cost using spongycastle.jce.provider package in Java running on android-based mobile phones, the effect of using different elliptic curves and analyze the security of both key agreements. Furthermore, we design a practical and efficient authentication mechanism to deploy our VoIP architecture and show that a VoIP call can be established in an acceptable interval. As a result, this paper provides a concrete and feasible architecture to secure a VoIP call.

Performance analysis of voice over internet protocol via non Markovian loss system with preemptive priority and server break down

Voice over Internet Protocol (VoIP) or Internet telephony has become an evolutionary and promising technology in the area of telecommunications. Lower costs and simpler maintenance are the major reasons of its increasing popularity. However, the network is inherently unreliable and highly challenged when it comes to quality of service (QoS). The VoIP system is primarily a multi-server system which may also face failures. Hence, in order to study the performance of VoIP, it is proposed to develop a multi-server, loss system with general service times, server breakdown and repair together with preemptive priority queuing discipline, for two types of customers: high priority and low priority. The customers are served immediately if the server is available. However, if all servers are busy, then on the arrival of a high priority customer, the service of a low priority customer is preempted. The steady state distribution of such underlying system is obtained together with the performance measures. A numerical illustration of the obtained results is also presented graphically. The results are validated using simulation.

Performance of VoIP in Wired-Cum-Wireless Ethernet Network

The Voice over Internet Protocol (VoIP) is a rapidly growing technology that enables transport of voice over data networks such as Ethernet Local Area Networks (LANs). This growth is due to the integration of voice and data traffic over the existing network infrastructure, low cost, and improved network management offered by the technology. This paper reports on the performance of VoIP traffic characteristics in a wired-cum-wireless Ethernet LAN. The effect of increasing the number of VoIP wireless clients, different voice codec schemes, and packet arrival distributions on system performance is investigated. Through various simulation experiments under realistic network scenarios, such as Small Office Home Office (SOHO) and campus networks, this paper provides an insight into the performance of VoIP over Ethernet LANs. Simulation results show that VoIP clients and voice codec schemes have significant effect on system performance. The authors preformed OPNET-based simulations to validate their experiments.

On the quality of VoIP with DCCP for satellite communications

SUMMARYWe present experimental results for the performance of selected voice codecs using Datagram Congestion Control Protocol (DCCP) with CCID4 congestion control over a satellite link. We evaluate the performance of both constant and variable data rate speech codecs for a number of simultaneous calls using the ITU E‐model. We analyse the sources of packet losses and additionally analyse the effect of jitter, which is one of the crucial parameters contributing to Voice over IP (VoIP) quality and has, to the best of our knowledge, not been considered previously in the published DCCP performance results. We propose modifications to the CCID4 algorithm and demonstrate how these improve the VoIP performance, without the need for additional link information other than what is already monitored by CCID4. We also demonstrate the fairness of the proposed modifications to other flows. Although the recently adopted changes to TCP‐Friendly Rate Control specification alleviate some of the performance issues for VoIP on satellite links, we argue that the characteristics of commercial satellite links necessitate consideration of further improvements. We identify the additional benefit of DCCP when used in VoIP admission control mechanisms and draw conclusions about the advantages and disadvantages of the proposed DCCP/CCID4 congestion control mechanism for use with VoIP applications. Copyright © 2012 John Wiley & Sons, Ltd.

An Optimization Technique for Improved VoIP Performance over Wireless LAN

The performance of wireless LANs is greatly affected by pathloss, RF interference and other sources of signal attenuation, in addition to network congestion. The primary factors involved in effective real-time communication, namely delay and loss, must be within certain controlled limits in such a scenario. In this paper, we analyze the various factors driving IEEE 802.11b access points through extensive simulations and thereafter develop an optimization technique to configure the parameters of the access points. We simulate our test bed scenario and apply the developed algorithm. We then implement the configured parameters in our SIP enabled test bed to provide optimum Voice over IP (VoIP) performance. Thereafter, we further analyze and optimize the parameters of the communicating nodes being realized through Xlite soft phones that enhance VoIP performance to a much better extent both for voice and video calls. Test bed results confirm an improvement from 22% packet loss to 3% considering both AP and node configurations together. Finally, we propose optimization algorithm for proper selection of threshold parameters in active queue management system. Results verify significant performance improvement in VoIP under congested scenarios. Simulation and measured results have also been included.

Efficient Control Channel Resource Allocation for VoIP in OFDMA-Based Packet Radio Networks

We propose an efficient control channel resource allocation approach to enhance the performance of voice-over-IP (VoIP) in orthogonal frequency division multiple access- (OFDMA-) based next generation mobile communication systems. As the long-term evolution (LTE) of universal terrestrial radio access network (UTRAN), evolved UTRAN (E-UTRAN) is the first OFDMA-based packet radio network and thus selected in this paper as an application example. Our proposed physical downlink control channel (PDCCH) resource allocation approach for E-UTRAN is composed of bidirectional power control, inner loop link adaptation (ILLA), and outer loop link adaptation (OLLA) algorithms. Its effectiveness is validated through large-scale radio system level simulations, and simulation results confirm that VoIP capacity with dynamic scheduling can be further enhanced with this PDCCH resource allocation approach. Moreover, the VoIP performance requirements for international mobile telecommunications-advanced (IMT-Advanced) radio interface technologies can be met with dynamic scheduling together with proposed PDCCH resource allocation. Besides E-UTRAN, this approach can be introduced to other OFDMA-based mobile communication systems for VoIP performance enhancement as well.

VoIP Performance over HSPA with Different VoIP Clients

High Speed Packet Access (HSPA) Holma H, Toskala A (in HSDPA/HSUPA for UMTS, 2006) is expected to provide enough bandwidth for voice over IP (VoIP) service. In this article we assess the performance of VoIP over HSPA with different VoIP clients and voice codecs. The simulations results show that VoIP can have a good voice quality over HSPA if a proper VoIP client and codec is used. However it is possible that the delay can increase with early HSPA implementations (mobile, network).

A performance study for IPv4–IPv6 translation in IP multimedia core network subsystem

AbstractA Next Generation Network (NGN) is an advanced, packet‐based network that exploits broadband and QoS‐enabled transport technologies for enabling multimedia services. In NGNs, the principles and requirements of IP Multimedia Sub‐system (IMS), which are used to deliver the desired benefits, should be carefully examined and studied. Internet Protocol version 6 (IPv6) is adopted by Third‐generation Partnership Project (3GPP) to solve address storage problem and provide new features (e.g. plug‐and‐play and mobility) for IMS. However, in the early stage of IPv6 deployment, the existing Voice‐over‐IP (VoIP) networks support Internet Protocol version 4 (IPv4) only. For IPv4–IPv6 interworking between IMS and the existing VoIP networks, the IMS‐Application Level Gateway (IMS‐ALG) and the Transition Gateway (TrGW) are proposed to translate Session Initiation Protocol (SIP) and Real‐time Transport Protocol (RTP) packets, respectively. In this paper, we focus on the IPv4–IPv6 translation for RTP packets, which is the bottleneck of VoIP performance. Specifically, we developed a TrGW called National Information and Communications Initiative (NICI)‐TrGW. In NICI‐TrGW, we perform IPv4–IPv6 translation at the Linux kernel and adjust the header room of kernel‐level packet buffer for each packet to reduce memory‐copy operations. We evaluate the performance of NICI‐TrGW and the existing solutions by using the SmartBits. Our study indicates that NICI‐TrGW outperforms the existing solutions in terms of three different output measures including packet loss rate, maximum throughput, and average latency. Copyright © 2009 John Wiley & Sons, Ltd.

Mobile-Taiwan experience in voice over IP-worldwide interoperability for microwave access trial

Considering voice as a dominant telecommunication service, the performance of Voice over IP (VoIP) plays a critical role in deployment of worldwide interoperability for microwave access (WiMAX) technology providing all-IP network services. To that effect, in this study, the authors investigate the performance of a WiMAX-based VoIP established under the mobile Taiwan (M-Taiwan) field-trial funded program. To achieve the objectives of the trial the measurement results expressed in the form of mean opinion score (MOS), packet loss, packet delay and jitters. For the worst-case scenario, the tests were conducted under a stringent condition of both communicating devices, wirelessly connected to the same WiMAX base station under a heavy background traffic and interference, were experiencing simultaneous handovers during the communication. Upon their analysis, the field measurements confirm an excellent performance when both communicating devices kept stationary and show an acceptable quality for the service when both communicating devices are on the move at a speed of 50 km/h.

Improving WLAN VoIP Capacity Through Service Differentiation

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%.

A Novel AP for Improving the Performance of Wireless LANs Supporting VoIP

Recently, wireless LANs based on the IEEE 802.11 standard have become popular and widely used, and Voice-over-IP (VoIP) is one of the most promising technologies to be used in mobile devices over wireless networks. Up-to-date, new technologies have improved, providing a high data rate to support various kinds of realtime applications. However, besides the need for higher data rates, seamless handoff is another important issue to be addressed in order to continue supporting real-time services across wireless LANs. In this paper we introduce a novel Access Point (AP) with two transceivers to improve network efficiency in terms of supporting seamless handoff and balancing the traffic load in a wireless network. In our proposed scheme, the novel AP will use the second transceiver to scan and find neighboring STAs in the transmission range and later send the result to its associate AP, which in turn compares and analyzes whether or not the STA should perform a handoff. The initial results from our simulations show the novel AP module is more effective than the conventional one in terms of providing a handoff process with low latency and support traffic load sharing with neighbor APs.

Assessing the quality of voice communications over internet backbones

As the Internet evolves into a ubiquitous communication infrastructure and provides various services including telephony, it will be expected to meet the quality standards achieved in the public switched telephone network. Our objective in this paper is to assess to what extent today's Internet meets this expectation. Our assessment is based on delay and loss measurements taken over wide-area backbone networks and uses subjective voice quality measures capturing the various impairments incurred. First, we compile the results of various studies into a single model for assessing the voice-over-IP (VoIP) quality. Then, we identify different types of typical Internet paths and study their VoIP performance. For each type of path, we identify those characteristics that affect the VoIP perceived quality. Such characteristics include the network loss and the delay variability that should be appropriately handled by the playout scheduling at the receiver. Our findings indicate that although voice services can be adequately provided by some ISPs, a significant number of Internet backbone paths lead to poor performance.