VoIP Performance Research Articles

VOIP Performance Over Broadband Wireless Networks Under Static and Mobile Environments

VOIP Performance Over Broadband Wireless Networks Under Static and Mobile Environments

Performance of VoIP Using Routing Open Shortest Path First with Multi-Protocol Label Switching

Voice over internet protocol is a communication technology in the world of computer network that can be used for sending voice or video and data transmission over Internet Protocol in real time. VoIP network can be implemented with Asterisk applications as a server to a Private Automatic Branch eXchange applied in a Graphical Network Simulator 3. In this study, VoIP communication using routing OSPF within MPLS will be calculated, the QoS value collected based on impact performance under normal condition and not normal condition (link failure) different varian bandwidth in the network. The results from the simulation show that in normal condition and not normal condition there is average delay value with routing OSPF 4 ms and routing OSPF with MPLS 5 ms, the value of jitter max which same of 6 ms using varian bandwidth from 256 kbps and 512 kbps. All of the QoS parameters, such as delay, jitter and packet loss will be compare to standard ITU-T G.114. This research can be extended with addition of another measurement or another protocol.

CODEC-BASED QOS ANALYSIS OF VOIP OVER CONVENTIONAL NETWORK AND SDN

Due to the advancement of technology in recent years, there has been a significant increase in the usage of bandwidth-consuming real-time applications such as VoIP, web-streaming. To cater to these ever increasing modern networking demands, Software-Defined Networking (SDN) is considered as the main structure for the future networking. We have analyzed and compared the QoS performance over conventional network and SDN, using Riverbed Modeler and Mininet Emulator, respectively, by considering eight different VoIP call scenarios based on the combination of voice codecs and call-type. The results for a single VoIP call scenarios showed 6 to 13 percent improvement in the throughput performance. In the case of concurrent VoIP calls, SDN can effectively handle congestion due to the efficient bandwidth utilization and the presence of a proactive centralized controller, resulting in improved QoS performance of VoIP in terms of throughput, jitter, packet loss and Mean Opinion Score (MOS).

The Road Not Taken: Re-thinking the Feasibility of Voice Calling Over Tor

Abstract Anonymous VoIP calls over the Internet holds great significance for privacy-conscious users, whistle-blowers and political activists alike. Prior research deems popular anonymization systems like Tor unsuitable for providing the requisite performance guarantees that real-time applications like VoIP need. Their claims are backed by studies that may no longer be valid due to constant advancements in Tor. Moreover, we believe that these studies lacked the requisite diversity and comprehensiveness. Thus, conclusions from these studies, led them to propose novel and tailored solutions. However, no such system is available for immediate use. Additionally, operating such new systems would incur significant costs for recruiting users and volunteered relays, to provide the necessary anonymity guarantees. It thus becomes an imperative that the exact performance of VoIP over Tor be quantified and analyzed, so that the potential performance bottlenecks can be amended. We thus conducted an extensive empirical study across various in-lab and real world scenarios to shed light on VoIP performance over Tor. In over half a million calls spanning 12 months, across seven countries and covering about 6650 Tor relays, we observed that Tor supports good voice quality (Perceptual Evaluation of Speech Quality (PESQ) >3 and one-way delay <400 ms) in more than 85% of cases. Further analysis indicates that in general for most Tor relays, the contentions due to cross-traffic were low enough to support VoIP calls, that are anyways transmitted at low rates (<120 Kbps). Our findings are supported by concordant measurements using iperf that show more than the adequate available bandwidth for most cases. Hence, unlike prior efforts, our research reveals that Tor is suitable for supporting anonymous VoIP calls.

Comparison of Voice over Internet Protocol (VoIP) Performances in Various Network Topologies

VoIP is a digital communication technology that is currently developing because VoIP can be implemented on several network topologies, such as bus, star, and ring. Each of these topologies has advantages and disadvantages. So, a study is required to find out in which topology can VoIP be implemented optimally. In this research, VoIP is implemented in several topologies and furthermore the performance measurements are carried out for each topology. VQ manager is installed in order to measure the VoIP performance. For the server, we used Elastix and for the node implementation network topologies, we use several access points. From the results of the research, the performance of VoIP implemented in the star topology produces QoS that is better than other topologies with a delay value of 185 ms, 18 ms jitter, and 1% packet loss. This happens because in the star topology, all packets are distributed centrally. It is expected that the results of this research can be used as a reference in the application of VoIP technology in several types of topologies.

Enhanced exponential rule scheduling algorithm for real-time traffic in LTE network

Nowadays, mobile communication is growing rapidly and become an everyday commodity. The vast deployment of real-time services in Long Term Evolution (LTE) network demands for the scheduling techniques that support the Quality of Service (QoS) requirements. LTE is designed and implemented to fulfill the users’ QoS. However, 3GPP does not define the specific scheduling technique for resource distribution which leads to vast research and development of the scheduling techniques. In this context, a review of the recent scheduling algorithm is reported in the literature. These schedulers in the literature cause high Packet Loss Rate (PLR), low fairness, and high delay. To cope with these disadvantages, we propose an enhanced EXPRULE (eEXPRULE) scheduler to improve the radio resource utilization in the LTE network. Extensive simulation works are carried out and the proposed scheduler provides a significant performance improvement for video application without sacrificing the VoIP performance. The eEXPRULE scheduler increases video throughput, spectrum efficiency, and fairness by 50%, 13%, and 11%, respectively, and reduces the video PLR by 11%.

Modelo Sip Seguro para una Comunicación extremo a extremo sobre IPV6

Modelo Sip Seguro para una Comunicación extremo a extremo sobre IPV6 Sip Security Model for End to End Communication on IPv6 Ross M. Benites, José L Quiroz, Raúl Villafani INICTEL-UNI, Lima 41 DOI: https://doi.org/10.33017/RevECIPeru2011.0024/ RESUMEN Las implementaciones VoIP hoy en día se han incrementado considerablemente, sin embargo no entodos los escenarios se tiene en cuenta los mecanismos de seguridad adecuados. Este último punto es muy importante a considerar el día de hoy , sobre todo por el agotamiento de las direcciones IPV4 y el despliegue hacia IPV6 de muchos de los servicios, donde aparecerán nuevas amenazas a la seguridad que trataran de opacar el gran auge de la tecnología VoIP. Si bien IPv6 fue desarrollado para solucionar muchas vulnerabilidades en seguridad que actualmente se ven presentes en IPv4, el hecho es que no logra alcanzar aún estas metas según pruebas realizadas. El protocolo SIP, el actor principal de la tecnología VoIP , requiere de la implementación de mecanismos de seguridad . Los escenarios actuales requieren terminales de usuario de alto rendimiento y soporte para adaptarse a mecanismos de seguridad heterogéneos o asumir relaciones de confianza. Sin embargo debemos tener en cuenta que hay varias combinaciones de soluciones de seguridad que son proporcionados por usuarios finales y los servidores. En este trabajo se expone un modelo de seguridad aplicado a un escenario experimental VoIP sobre el Internet de Próxima Generación (IPv6), que utiliza la seguridad salto a salto y extremo a extremo. El escenario propuesto se encuentra sobre una red local con direcciones IPv6. Utiliza dos servidores Asterisk implementados bajo las mismas características que cumplen la función de SIP Proxy, y se encuentran conectados mediante un enlace troncal SIP – TLS. Se utilizan además dos terminales de usuario (teléfonos IP) provenientes de una marca comercial conocida, registrados cada uno mediante el protocolo SIP-TLS a cada servidor Asterisk. En trabajos anteriores, se han realizado varios estudios sobre el rendimiento del uso de VoIP sobre IPv4 e IPv6 comparando los resultados [1], evaluación de mecanismos de seguridad para mantener la autenticación de usuario, confidencialidad e integridad de la señalización y media de los mensajes VoIP sobre las redes IPv4 [2] y [3]. Este trabajo se esboza en un marco de seguridad, donde se presenta un escenario basado en una red VoIP en IPV6 utilizando TLS y SRTP. TLS es utilizado para la seguridad en el establecimiento de la sesión con mecanismos de autenticación salto a salto y SRTP (Secure Real Time Protocol) para la seguridad del establecimiento del stream de media. Nos enfocaremos en analizar y evaluar la seguridad de los mensajes en este escenario sobre el protocolo de transporte seguro (TLS). Descriptores: sip, tls ,ipv6, srtp. ABSTRACT VoIP deployments today have increased considerably, but not all the scenarios consider appropriate security mechanisms. This last point is very important to consider today, especially the depletion of IPv4 addresses and the deployment of many services IPv6, where will new security threats to try to overshadow the great technology boom VoIP. Although IPv6 was developed to solve many security vulnerabilities are currently present in IPv4, the fact is that still fails to achieve these goals by testing. The SIP protocol, the main actor of VoIP technology requires the implementation of security mechanisms. The current scenarios require high-end user performance and support to adapt to heterogeneous security mechanisms or assume trust relationships. But keep in mind that there are various combinations of security solutions that are provided by end users and servers. This paper presents a security model applied to an experimental scenario VoIP over Next Generation Internet (IPv6), which uses hop by hop security and end to end. The proposed scenario is on a local network with IPv6 addresses. Use two Asterisk servers implemented under the same characteristics that act as SIP Proxy, and are connected via SIP trunk - TLS. They also use two user terminals (IP phones) from a known trade mark registered by each SIP-TLS protocol for each server Asterisk. In previous work, there have been several studies on the performance of VoIP using IPv6 and IPv4 and comparing the results [1], evaluation of security mechanisms to support user authentication, confidentiality and integrity of the signaling and media messages VoIP over IPv4 networks [2] and [3]. This paper outlines a framework of security, which presents a scenario based on a VoIP network in IPv6 using TLS and SRTP. TLS is used for the security session establishment authentication mechanisms hop by hop and SRTP (Secure Real Time Protocol) for the safety of the establishment of media stream. We will focus on analyzing and evaluating the security of the messages in this scenario the secure transport protocol (TLS) . Keywords: sip, tls ,ipv6, srtp.

Computer Network Simulation of Firewall and VoIP Performance Monitoring

Fast growing in communication technology has influenced global changes and challenges appear in the field of network security issues. Security solutions must be efficient and operate in a way to deal with threats, reject and stop the network intruders and Trojans. The simulated network of Salahaddin university new campus is planned to build on an area of (3000X3000) meter square. The network consists of many main and secondary devices. The network is mainly consisted of one core switch that provides a very high data transfer through connecting all the collected positions by a variety of cable medias to the entire network switches which are installed in each college location. The department of academic administration (presidency) of the university design is similar to the network designed in each college. The mentioned switch obtains the services from a router that isolates the network from the cloud which supports the services of internet to the network. The firewall is connected to the switch that connects the main server and cloud together. This paper focuses on undertaking a simulation to analyze and examine the performance of the whole network when two scenarios are implemented, the first is if firewall devices is used, the second is when firewall is not used, since the project of building Salahaddin University new campus is at the initial stage, therefore, the researchers think that, it is very important to figure out the drawbacks and deadlocks of using firewall upon each branch of the network and overall network performance before the submitting the final networks design that going to be implemented and installed, because this will indicate many differences on the construction, for example, the network panels ways, the cable collecting locations, network channels and many other device fixing things depending on the media types in addition of the demand of future expansion capabilities. The results show that the using or adding of firewall device to the university campus computer network, will improve the overall network performance though increasing the data stream on many network sections and sectors, also better results are obtained.

Comparative Analysis Study on Network Performance of VoIP and Data Traffic Pattern using Queuing Algorithms in Wired Network Environment: An Implementation using OMNeT++ Simulator

Objectives: To improve the performance of network traffic using different queuing techniques in wired network environment. Methods/Statistical Analysis: We have simulated Quality of Service and Queuing algorithms to test the performance evaluation of VoIP data traffic in wired networks. The network features are taken as the input and applied in simulation to test the data traffic in wired network environment. We have used OMNET simulator to run the designed scenarios. The simulator is scalable to real time simulation, network emulation, database support and integration. One router was implemented in with queuing techniques to determine how the packets are buffered during waiting time of transmission and results were analysed and explained in detail. Findings: The experimental results show that QoS is a better technique to control and manage reliable communication in wired networks. It is very important that network administrator to apply appropriate queuing and policing techniques to improve the performance of the network. One of the observations made is that Priority Queuing is better than First in First Out technique to manage the traffic load and performance of the network. In another set of results, First in First Out technique is good for normal network not for traffic network. FIFO performance is better compared with PQ in normal networks. Application/Improvements: Experiments were performed through implementing scenarios in OMNet++. Performance evaluations were carried out and explained based on measuring queue length, queuing time, and packets received an end to end delays with examples. OMNET++ have several limitations due to it open source; therefore, in future we are planning to implement the scenarios on real hardware to compare the results difference between simulations and real hardware. Keywords: Quality of Service (QoS), Queuing, VoIP, Wired Network

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.

The Performance Evaluation of Voip for a Mobile User

Broadband Wireless Access (BWA) technology has developed rapidly and it has dominated the advancement of technology especially in wireless networks. This paper studies the performance evaluation of VoIP for a mobile user and how the QoS parameters vary for different speeds. The purpose of the study is to examine the effect of mobility of a user on the performance of VoIP. In the meanwhile, it is evaluating the simulation and emulation method for its validity to be used for any new upcoming technology. However, the results of the simulation need to be verified and assessed by emulation tested results, hoping that both results match closely with each other.

VoIP Call Admission Control in WLANs in Presence of Elastic Traffic

VoIP service over WLAN networks is a promising alternative to provide mobile voice communications. However, several performance problems appear due to i) heavy protocoloverheads, ii) unfairness and asymmetry between the uplink and downlink flows, and iii) the coexistence with other traffic flows. This paper addresses the performance of VoIP communications with simultaneous presence of bidirectional TCP traffic, and shows how the presence of elastic flows drastically reduces the capacity of the system. To solve this limitation a simple solution is proposed using an adaptive Admission and Rate Control algorithm which tunes the BEB (Binary Exponential Backoff) parameters. Analytical results are obtained by using an IEEE 802.11e user centric queuing model based on a bulk service M=G[1;B]=1=K queue, which is able to capture the maindynamics of the EDCA-based traffic differentiation parameters (AIFS, BEB and TXOP). The results show that the improvement achieved by our scheme on the overall VoIP performance is significant.

Capability Measure for VoIP Performance

Abstract Voice-over Internet protocol (VoIP) has become more popular because of the widespread use of the Internet. The quality of VoIP phone calls is typically not as stable as that in the public switched telephone network. This is because of the packet-switching approach of IP networks. This situation is very common for a mobile device that connects a VoIP call through a wireless link. Therefore, properly managing the quality of service for VoIP calls is important. Some objective methods are developed to evaluate subjective qualities. They are based on the physical system. This study is based on statistical quality control and the process capability index to evaluate VoIP's performance, then demonstrates the entire system's service quality. The acceptable thresholds of two critical characteristics in voice-quality transformation, including packet loss and delay, are regarded as processing specifications. Our method measures the process capability of every phone call item to meet processing specifications where multiple phone call quality characteristics are involved.

The Good, the Bad and the WiFi: Modern AQMs in a residential setting

Several new active queue management (AQM) and hybrid AQM/fairness queueing algorithms have been proposed recently. They seek to ensure low queueing delay and high network goodput without requiring parameter tuning of the algorithms themselves. However, extensive experimental evaluations of these algorithms are still lacking. This paper evaluates a selection of bottleneck queue management schemes in a test-bed representative of residential Internet connections of both symmetrical and asymmetrical bandwidths as well as WiFi. Latency under load and the performance of VoIP and web traffic patterns are evaluated under steady state conditions. Furthermore, the impact of the algorithms on fairness between TCP flows with different RTTs, and also the transient behaviour of the algorithms at flow startup is examined. The results show that while the AQM algorithms can significantly improve steady state performance, they exacerbate TCP flow unfairness. In addition, the evaluated AQMs severely struggle to quickly control queueing latency at flow startup, which can lead to large latency spikes that hurt the perceived performance. The fairness queueing algorithms almost completely alleviate the algorithm performance problems, providing the best balance of low latency and high throughput in the tested scenarios. However, on WiFi the performance of all the tested algorithms is hampered by large amounts of queueing in lower layers of the network stack inducing significant latency outside of the algorithms’ control.

The Effects of Different Congestion Management Algorithms over Voip Performance

This paper presents one of the features of DS (Differentiated Services) architecture, namely the queuing or congestion management. Packets can be placed into separate buffer queues, on the basis of the DS value. Several forwarding policies can be used to favor high priority packets in different ways. The major reason for queuing is that the router must hold the packet in its memory while the outgoing interface is busy with sending another packet. The main goal is to compare the performance of the following queuing mechanisms using a laboratory environment: FIFO (First-In First-Out), CQ (Custom Queuing), PQ (Priority Queuing), WFQ (Weighted Fair Queuing), CBWFQ (Class Based Weighted Fair Queuing) and LLQ (Low Latency Queuing). The research is empirical and qualitative, the results are useful both in infocommunication and in education.

데이터 트래픽 중심의 LTE망에서 VoIP를 위한 스케줄링 알고리즘 성능 분석

최근 국내외에서는 LTE가 도입 초기단계를 넘어, LTE-A로의 도약이 이루어지고 있다. 이를 기반으로 고속의 서비스를 제공함으로써, 이동 전화로 시작한 휴대폰의 역할은 데이터 중심의 스마트폰으로 바뀌고 있다. 시스코에 의하면 2018년 전세계 모바일 데이터 트래픽은 지난해 대비 11배 증가 전망할것으로 내다봤다. 한국에서 모바일 비디오가 전국 모바일 트래픽에서 차지하는 비중은 2013년 66%에서 2018년 75%로 증가할 전망이다. 하지만 여전히 음성 서비스가 가장 중요하며, 높은 VoIP 통화 품질을 제공하기 위해서는 다양한 데이터 트래픽을 사용하는 가운데 LTE 망 상황에 따라 전송 속도를 조절하고, 낮은 BLER을 제공해야 한다. 실제 통화를 위한 전송율이 최대 23.85 Kbps인 AMR-WB 코덱을 사용하지만, 초당 수십~수백 MB까지 전송할수 있는 LTE일지라도 셀 가장자리의 사용자에게 일정 수준의 전송률을 떨어뜨리지 않고 유지하는 것은 어려운 일이다. 본 논문에서는 이러한 점을 개선하기 위해 특히 문제가 많이 발생하는 셀 가장자리의 품질을 개선하고자, 다양한 스케줄링 알고리즘의 성능을 분석하고 개선 방안을 제시하였다. 제시된 방안은 2계층 스케줄링 알고리즘을 사용하며, 셀 가장자리에 위치한 VoIP 사용자를 일정 비율 타 어플리케이션에 비해 우선 처리하는 스케줄링 알고리즘을 제공하는 기술이다. 본 논문에서는 throughput과 BLER 측면에서 VoIP 성능이 향상되었음을 확인하였다.

Identifying traffic differentiation on cellular data networks

The goal of this research is to detect traffic differentiation in cellular data networks. We define service differentiation as any attempt to change the performance of network traffic traversing an ISP's boundaries. ISPs may implement differentiation policies for a number of reasons, including load balancing, bandwidth management, or business reasons. Specifically, we focus on detecting whether certain types of network traffic receive better (or worse) performance. As an example, a wireless provider might limit the performance of third-party VoIP or video calling services (or any other competing services) by introducing delays or reducing transfer rates to encourage users to use services provided by the wireless provider. Previous work explored this problem in limited environments. Glasnost focused on BitTorrent in the desktop/laptop environment, and lacked the ability to conduct controlled experiments to provide strong evidence of differentiation. NetDiff covered a wide range of passively gathered traffic from a large ISP but likewise did not support targeted, controlled experiments. We address these limitations with Mobile Replay.

Performance Analysis of VOIP Traffic Over Integrating Wireless LAN and WAN Using Different Codecs

A simulation model is presented to analyze and evaluate the performance of VoIP based integrated wireless LAN/WAN with taking into account various voice encoding schemes. The network model was simulated using OPNET Modeler software. Different parameters that indicate the QoS like MOS, jitter, end to end delay, traffic send and traffic received are calculated and analyzed in Wireless LAN/WAN scenarios. Depending on this evaluation, Selection codecs G.729A consider the best choice for VoIP.

Audio Codecs in VoIPv6: A Performance Analysis

Audio communications in IP based networks have been revolutionized by the introduction of VoIP applications. High cost-efficiency has made VoIP to be the communication means in today’s world; and this trend is anticipated to be continued on an ongoing basis. The performance of VoIP significantly depends on the efficiency of the audio codecs used in any communication scenario which make the study on the performance issues of audio codecs in VoIP applications worth investigating. IPv6 is the new version of IP, which will gradually replace the current IPv4 as the transition from IPv4 to IPv6 is already in place. This demands the scrutiny of the audio codecs being used in IPv4 to be tested for their compatibility in IPv6 in terms of desired performance. This paper presents the study on the performance of selected audio codecs that are widely used in VoIPv4. G.711, G.729A and G.723.1 codecs were chosen for the study in VoIPv6 based scenarios presented in this paper. The selected audio codecs were applied in IPv6 based voice communication network scenarios to determine their performance efficiency by observing various QoS parameters. The study was done by means of simulation using OPNET.

An IP-based packet test environment for TD-LTE and LTE FDD

In recent years, Long Term Evolution systems have been developed to replace 3G mobile networks. During LTE network deployment, it is essential that mobile operators and manufacturers conduct tests on user equipment and the base station (eNB). Besides conformance testing at the radio layer, it is also important to conduct network-layer performance tests between UE and eNB. This article develops an LTE test environment that can be used to test the IP-based packet performance for TD-LTE and LTE FDD. Specifically, we consider the latency and the throughput performance of IP, TCP, and VoIP (i.e., G.711) packet delivery. This multi-purpose test environment can be utilized to investigate the performance of the UE and/or eNBs of various manufacturers. Furthermore, we use this environment to compare TD-LTE and LTE FDD. The measurement results provide guidelines for UE and eNB vendors to improve their products, and operators to deploy their LTE networks and subsidized UE devices.