VoIP Service Quality Research Articles

Analysis of the Impact of Loops or Flapping in VoIP Quality of Service

Analysis of the Impact of Loops or Flapping in VoIP Quality of Service

Impact Analysis of Insufficient Destination Resources on VoIP Service Quality: Case Study

Impact Analysis of Insufficient Destination Resources on VoIP Service Quality: Case Study

USE OF THE STATISTICAL ANALYSIS METHODS TO DETECT VOIP NETWORK TRAFFIC ANOMALIES

The purpose of the study is to automate detection of signs of technical malfunction and deterioration in the quality of services in the VoIP network. Deterioration of quality leads to a decrease in the volume of services provided, and consequently a profit, a decrea se in customer loyalty and loss of market share. There are three levels of quality degradation in VoIP networks: lack of access at the data network layer, inability to establish a voice connection and lack of access at the level of application services, and degradation of service quality. The analysis of absolute and relative statistical quality indicators of IP-telephony in accordance with the requirements of the international ITU-T standards has been performed, and main indicators and formulas for their calculation are listed. The values of statistical quality indicators are periodically calculated for external channels, subscriber groups and tariff directions. The primary data is CDR (call data records). To detect anomalous changes in the values of the quality indicators it is proposed to use the exponential smoothing method – the Holt-Winters additive model. The deviation of the current value of the quality indicators from the forecasted one is calculated. The confidence range is calculated using the Brutlag method. If the deviation goes beyond the confidence range, the change in the value of the indicator is considered anomalous and the value of the anomaly coefficient of the indicator is set. The period is characterized by a vector of anomaly coefficients of all quality indicators. To classify the period as anomalous, the value of the module of the anomaly coefficients vector of the given period is used. The features of the method application are also considered, in particular the choice of the seasonal period and the calculation of anomaly coefficients during periods of minimum load. The proposed method allows to diagnose an abnormal change in the values of VoIP service quality indicators in automated mode. The prototype of an automated system for the quality of VoIP service monitoring was developed on the base of the method described.

Queuing Impact on VoIP Quality of Service (QoS) in Telecom Networks

Queuing Impact on VoIP Quality of Service (QoS) in Telecom Networks

A Comprehensive Study of DSCP Markings’ Impact on VOIP QoS in HFC Networks

Various factors can have a significant degrading impact on the residential Voice over Internet Protocol (VoIP) phone services’ quality. Hybrid fibre- coaxial (HFC) networks typically carry three types of traffic that include voice, data, and video. Unlike data and video, some delays or packet loss can result in a noticeable degraded impact on a VoIP’s phone conversation. This paper will analyze and assess VoIP traffic prioritization and its impact on VoIP’s quality of service (QoS) based on the concept of differentiated services code point (DSCP) markings. Call testing examines two types of calls. The first set of tests focus on calls that originate from a VoIP network and terminate on a signalling system 7 (SS7) network. The second experiment focuses on calls that originate from SS7 network and terminate on a VoIP network. The research results provide DSCP markings configurations that can improve phone conversations’ quality.

Análisis de la Arquitectura DIFFSERV sobre redes MPLS para la provisión de QoS en aplicaciones en tiempo real (VoIP)

Actualmente, las redes IP necesitan de QoS para aplicaciones en tiempo real, donde el principal problema está en que las redes IP no tienen un mecanismo para proporcionarlo. En este trabajo, se propuso implementar la arquitectura DiffServ sobre redes MPLS, con el objetivo de mejorar las prestaciones a las redes, brindando un trato diferenciado al tráfico en tiempo real. En el análisis se diseñó un ambiente de pruebas con tres escenarios: redes IP, redes MPLS y MPLS DiffServ, realizados en GNS3 con la herramienta Dig-it 2.61 GUI 0.92, para inyectar tráfico de VoIP y datos utilizando etiquetas para su identificación. Cada contexto fue evaluado con los tres indicadores: retardo, jitter y pérdida de paquetes, mediante la herramienta Wireshark. Nuestro estudio determinó una calidad de servicio de VoIP adecuada con la implementación de MPLS y DiffServ.

Allocation For Wi-Fi Devices At Campus of Udayana University in Bukit Jimbaran Using Atoll Applications

Access to intranet and internet networks in the campus area requires a number of Wi-Fi devices to cover the coverage area of Udayana University. The Wi-Fi service is expected to be used for VoIP applications in the context of wireless communications. The IEEE 802.11g standard specifies an outdoor service range of 140 m and an indoor service range of 38 m. From research conducted in the engineering faculty area, the quality of VoIP service uses Wi-Fi with range of 25 m. In this research, the design of Udayana University campus Wi-Fi needs using Atoll application with IEEE802.11g standard at 140 m, 38m and 25m ranges. The results obtained to cover Udayana University campus area require Wi-Fis: 7 units for range 140 m, 98 units for range of 38 m and 225 units for range of 25 m

Design of PBX IP VoIP Server in State Polytechnic of Malang Network Based on Single Board Circuit

Building VoIP technology requires a server called Elastix. Singleboard circuit device can be an alternative to cover the shortcomings of a PC as a VoIP server, to support practicum activities. The specifications for the quad-core processor and 1GB LPDDR2 memory are expected to be able to provide access to the latest version of the operating system to be able to produce better communication quality as I PPBX. This study designed a VoIP IP PBX server communication system at the Polinema Telecommunication Laboratory using the Elastix Linux operating system, to determine the quality of VoIP communication services and the performance of a single board circuit as an IPPBX server. The results of the research are 64 kbps bandwidth capacity capable of serving a customer capacity of 181 subscribers through 8 number of channels. QoS measurement obtained a delay value of 4.87091 ms; jitter of 18,265ms; packet loss of 0.71%; throughput of 0.2747 Mbit / sec. The singleboard circuit server is capable of serving VoIP communications for as many as 20 users and 10 calls simultaneously on the same server with a cpu usage percentage of 17.7% and 17.2% cpu usage, when communicating with other servers.

Evaluating the Impact of Routing on QoS of VoIP over MANET Wireless Networks

In this paper, a performance analysis study for VoIP over mobile wireless ad hoc networks (MANET) is carried out using OPNET tool. The AODV, DSR, TORA, OLSR, and GRP routing protocols have been used as candidates to evaluate the impact of routing on quality of service (QoS) of VoIP application over MANET networks. Three different queuing mechanisms (FIFO, PQ, and WFQ) have been used also as nominee of queuing techniques. The VoIP is applied and the QoS is measured in terms of jitter delay, end-to-end packet delay, and wireless LAN media access delay and wireless throughput of 802.11 g technology with 54 Mbps.

Yagi Antenna Design and Analysis 11 Folded Dipole Feeding Elements for VoIP Networks

The problem that often occurs in VOIP networks using wireless media is the weak signal received and transmitted by the transceiver device at a long distance from the wireless router, so that the quality of VOIP service is often disconnected. The purpose of this research is to design and realize an 11 element yagi antenna with folded feed elements. dipole as an antenna for a transceiver device whose gain is better than a built-in antenna, so that it can support VOIP services. Yagi antenna is an antenna consisting of 3 elements, namely a reflector element, a driven element and a director element. The reflector element and the director element cause the yagi antenna to be directional so that a large gain is obtained from a certain point, while the driven one is used is a folded dipole antenna which has an impedance of 300 ohms so that the matching technique to the 50 ohm transmission line becomes easier. Ghz is less than 1.5 and RL is less than -14 dB. The bandwidth obtained is 620 MHz, the gain is 16.08 dBi with directional radiation and linear polarization. In implementation, the VOIP service is stable at a distance of 50 m using a yagi antenna, compared to the less stable built-in antenna.

Research on Data Packets Clustering Algorithm in the Wireless Multiple Hop Network

In view of the problems existing in the wireless multiple hop network such as consumption imbalance of node power, disunity of node transmission data efficiency, unfixed life within the scope of network, it has put forward the trade-off relationship between the wireless multiple hop network node energy consumption and multiple factors based on the k-means clustering method. The principle of steps and characteristics of the k-means clustering algorithm are first introduced; Then model the influence of the K-means polymerization on VoIP service quality, then use the k-means clustering method to make cluster analysis for network node data package, and mine the trade-off relationship between data transmission service quality and multiple hops node energy consumption; Finally carry on the simulation experiment to test the performance of this method. Simulation results show that the method not only improves the data transmission service quality of VoIP service, but also reduces the energy consumption of nodes and prolongs the life span of the wireless network.

Cross-Layered Secure and QoS Aware Design of VOIP over Wireless Ad-Hoc Networks

In this paper, Cross-layer design has been used to provide quality of service (QoS) and security at the same time for VOIP over the wireless ad-hoc network. In this paper the authors extend their previous work (i.e. Multi-path Multi-Channel Protocol Design for Secure QoS-Aware VOIP in Wireless Ad-Hoc Networks) by adding transport and application layers considerations. The goal of this paper is to support QoS and security of VOIP simultaneously. Simulation results shows that the proposed cross-layered protocol stack design significantly improve QoS parameters of the VOIP calls under the jamming or Denial-of-service attacks.

Comparative Analysis of the Performance of Different Codecs in a Live VoIP Network using SIP Protocol

In this paper we analyzed the impact of different codecs on the quality of VoIP services. We analyzed the performance of VoIP networks to have made an assessment to what extent are fulfilled the expectations regarding the desired quality of service and the authors applied the following methodology: we selected according to certain criteria and the same codecs implemented in a VoIP network, then selected the appropriate traffic routes and measurements carried out on them, and finally at the end is based on measurement results obtained comparative analysis and the comparisons made packet loss parameter. In the paper are discussed the main problems regarding the quality of VoIP services: and the choice of coding techniques. Measurements were performed with common used codecs for VoIP: G.711, G.723 and G.729, and the variation values of packet size and number of calls. Also, measurements were performed with optical and wireless transmission system. Finally, the main objective of this study was to select an appropriate codec to get a high quality voice and tries to compromise between price/quality of service. In conclusion, G.711 is the preferred technique that provides high voice quality (MOS greater than 4) but with a flaw need high bandwidth, and G.723 provide relatively good voice quality (MOS between 3 and 4) with a much smaller bandwidth, which means it can handle multiple users. The G.729 has a high level of compression and proper technique for many users, but only when the voice quality is not taken into account. Ill. 11, bibl. 9, tabl. 1 (in English; abstracts in English and Lithuanian).DOI: http://dx.doi.org/10.5755/j01.eee.117.1.1050

Evaluation of VoIP Service Quality under the Roaming of Mobile Terminals

In this paper we evaluated by simulation the effect of VoIP roaming and data traffic roaming on other VoIP calls. Regardless of MIPv4 or MIPv6, the quality of voice of all VoIP calls falls down quickly to the bottom level after the start of roaming by one voice terminal. That was caused by the excessive retransmission on downlink. Thus it seems that we need a kind of call admission control when we adopt the roaming service on VoIP calls. Data traffic degrades also the voice quality especially at the foreign agent side.

VoIP Quality Measurement System Using Flow Mediation for Large-Scale IP Networks

We present the development of a VoIP quality of service (QoS) measurement system that enables operators to diagnose a QoS degradation segment. Our system uses a flow-based passive measurement method to fulfill the requirement for QoS measurement in large-scale IP networks. In particular, we adopt an access control list (ACL)-based filtering function that selects traffic to monitor and develop a function for correlating signals and media data records. This correlation function is required to dynamically configure ACL-based filtering for monitoring media streams whose port numbers are determined by a signaling protocol. To improve the scalability of existing measurement systems, we also develop a hardware-based filtering engine on a commercial switch as well as a mediation box that performs QoS calculation based on traffic records exported by the engine in a distributed manner. We demonstrate the feasibility of the measurement system by evaluating a prototype system.

Study on VoIP Service Quality Management

VoIP transmits voices over IP-based networks and it is the abbreviation of Voice over Internet Protocol. Recently, VoIP provides various services in addition to voices. Since VoIP services' provision is extending, VoIP service quality management is becoming an important issue. Therefore, in this paper, we study VoIP service quality management. We examine VoIP technology, service types, and network architecture. Then, we investigate key quality indicators(KQIs)/key performance indicators(KPIs) in terms of customers, not network service providers. Toward this, we also study the concept of general service quality management as well as the concept of telecommunication related service quality management. Moreover, we apply and R charts to show how to use statistical quality control techniques in real telecommunication companies with one KQI.

Mobile WiMAX에서 IPTV 및 VoIP 음성서비스 품질을 고려한 수면구간 길이와 지터버퍼 크기의 상관관계 분석

IPTV 및 VoIP 서비스는 높은 이동성과 전송 속도를 보장하는 Mobile WiMAX 네트워크 상에서 제공할 수 있는 유용한 응용 서비스들이다. IPTV의 오디오 전송이나 VoIP의 통화 품질에 영향을 미치는 요소 중 전송 경로의 잦은 변경이나 경로간 전송 시간의 차이에 따라 발생하는 지터에 의한 패킷 손실은 지터 버퍼를 이용하여 완화할 수 있다. 본 논문에서는 Mobile WiMAX 네트워크 상에서 이동 단말의 전력 소모 절감을 위해 사용되는 PSC-II 모드를 사용할 때의 오디오 및 음성 서비스의 품질(Quality of Service)과 지터 버퍼 크기의 상관관계에 대해 연구, 분석한다. 이를 위해 절전 모드 사용으로 인해 추가로 발생하는 지연 시간을 포함한 서비스의 종단간 지연시간 모델과 종단간 지연시간을 기준으로 한 서비스 품질 기준을 제시하였다. 또한, 제시한 모델의 다양한 파라미터에 따른 시뮬레이션 분석 결과를 통해 절전 모드를 사용할 경우에는 지터 버퍼의 크기 증가에 따른 지연으로 인한 패킷 손실이 오히려 오디오 및 VoIP 서비스 품질 측면에서 좋지 않은 영향을 미칠 수 있음을 보였다. IPTV and VoIP services are considered as killer applications over Mobile WiMAX network, which provideshigh mobility and data rate. Among those which affect the quality of voice in those services, the jitter buffer or playout buffer can compensate the poor voice quality caused by the packet drop due to frequent route change or differences among routes between service endpoints. In this paper, we analyze the correlation between the sleep interval length and jitter buffer size in order to guarantee a predefined level of voice quality. For this purpose, we present an end-to-end delay model considering additional delay incurred by the WiMAX PSC-II sleep mode and a VoIP service quality requirement based on the delay constraints. Through extensive simulation experiments, we also show that the increase of jitter buffer size may degrade the voice quality since it can introduce additional packet drop in the jitter buffer under WiMAX power saving mode.

Analysis of Optimal Jitter Buffer Size for VoIP QoS under WiMAX Power-Saving Mode

VoIP service is expected as one of the key applications of Mobile WiMAX, but the speech quality of VoIP service often suffers deterioration due to the fluctuating transmission delay called jitter. This is commonly ameliorated by a de-jitter buffer, and we aim to find the optimal size of de-jitter buffer to achieve speech quality comparable to PSTN. We developed a new model of the packet drops at the de-jitter buffer and the end-to-end packet delay which takes account of the additional delay introduced by the WiMAX power-saving mode. Using our model, we analyzed the optimal size of the de-jitter buffer for various network parameters, and showed that the results obtained by analysis accord with simulation results.

A Policy-Based Admission Control Scheme for Voice over IP Networks

Problem statement: In Voice Over IP (VOIP) network, when more calls are admitted to the network, more voice packet traffic is created. Since bandwidth is always limited, this may result network congestion and/or may affect voice quality. Thus, we needed a mechanism for improving the Quality of Service (QoS) by controlling VOIP calls admission. Approach: Given a specified bandwidth and a constant background data rate, we attempted to explore the effect of Open Window and Leaky Bucket admission schemes on VoIP calls quality. These policy-based admission controls were simulated using NS-2 Simulator. The inter-arrival time distribution for the network background data traffic was assumed to be deterministic with a Constant Bit Rate (CBR). Voice packets traffic inter-arrival time is assumed to have an exponential distribution. Each voice call has a rate of 64 kbps for duration of 120 min. Results: Various performance measures of VoIP calls and packet traffic were evaluated including: packet loss, packet drop rate, delay, jitter and call rejection rate. Performance results of the experiment are summarized in a power ratio index which presented the impact of a collection of performance parameters on VoIP service quality. Conclusion: Implementing a policy based admission scheme on VoIP network will improve its QoS and the degree of improvement depends on the network setting parameters. If threshold rate for call admission is set above network ceiling bandwidth, leaky bucket will result a higher and unacceptable jitter. Overall, leaky bucket scheme was considered inferior when compared to open window for improving QoS of VoIP.

VoIP Quality of Service in Networks with Active Queue Management

This article provides a new analytical method based on a fixed point approach to estimate the impact of active queue management schemes on the quality of service of voice over IP applications. The results obtained by the proposed method were validated using discret event simulation techniques. In all the studied cases it was observed a great deal of agreement between the analytical results and the results obtained through simulation. The theoretical predictions, as well as the presented empiric evidences demonstrate that the use of active queue management offers a better quality of service than the traditionally queue control mechanisms used in internet.