VoIP 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

Performance Analysis of G.711 and G.729 Codec Schemes under Various Queuing Techniques in Voice over Internet Protocol Transmissions

Advancements in internet technology have enabled the integration of different traffic types i.e. data, video, and voice into a single network. This technology offers many benefits but also presents some challenges. Real-time traffic services such as VoIP require a certain Quality of Service (QoS) which cannot be guaranteed on the Internet therefore, key performance metrics become all the more important. The choice of codecs and queuing techniques becomes crucial for ensuring optimal performance, especially in networks with diverse traffic types. This research therefore compared the effects of the combinations of these tools (i.e. queuing techniques and codec schemes) on the quality of VoIP. A simulation approach using the OPNET Modeler 14.5 tool has been used to simulate a network supporting three different types of traffic namely: FTP traffic, Video conferencing traffic, and VoIP traffic. While maintaining the same topology and traffic of the network, different types of codec schemes and queuing techniques have been tested through the measurement of parameters such as delay and throughput. The custom Queuing technique showed the best performance overall while FIFO suffered the highest delay. The graphs were observed to follow the same pattern regardless of the codec scheme used however, G729 performed the better of the two as it received higher amounts of voice traffic and slightly lower delays compared to G711.

Enhancing VoIP quality in mobile networks through optimised MOS prediction and base station clustering with SDN

Enhancing VoIP quality in mobile networks through optimised MOS prediction and base station clustering with SDN

Enhancing VoIP Quality in Mobile Networks through Optimized MOS Prediction and Base Station Clustering with SDN

Enhancing VoIP Quality in Mobile Networks through Optimized MOS Prediction and Base Station Clustering with SDN

Implementing a campus VoIP intercom VLAN: A technology review, system requirements and architecture

Voice over the internet can be used with existing Local Area Network infrastructure within an organisation such as DIT to cut down the communication costs within the campus premises. This paper presents how can VoIP intercom be set and the requirements of the system. Voice is very important method of communication and hence is important to make it cheap and easily available. In this work, a VLAN was created in the DIT LAN then an appliance IP-PBX was installed, the phones were connected to a nearest by backbone switch. With gateway of the phones being the IP-PBX and VLAN tagging of the respective connected switch ports, the phones were able to connect to the PBX. The analysis of the system during busy hour shows that the system reliable and suitable to be used if the concurrent calls do not exceed the maximum capacity of the PBX. It is on this basis that this paper tends to critically review this new technology VoIP, x-raying the different types. It further more discusses in detail the VoIP system, VoIP protocols, and a comparison of different VoIP protocols. The compression algorithm used to save network bandwidth in VoIP, advantages of VoIP and problems associated with VoIP implementation were also critically examined. It equally discussed the trend in VoIP security and Quality of Service challenges. It concludes by reiterating the need for a cheap, reliable and affordable means of communication that would not only maximize cost but keep abreast with the global technological change.

VoIP Performance Evaluation and Capacity Estimation Using different QoS Mechanisms

Data networks and new mobile networks (4G and 5G) deploy packet switching in their core networks. Due to this architecture VoIP protocols has a wide deployment in these environments. VoIP capacity and quality must be considered during VoIP protocols implementations. The aim of QoS mechanisms is to satisfy voice traffic requirements; this is deployed by using many tools like congestion management utilities, congestion avoidance utilities, and link fragmentation tools. Each tool has an impact on voice performance. Queuing is very important mechanisms in the traffic management system. Certain routers in data networks must deploy some QoS tools that control how different packets are temporally buffered until transmission on the interface. This paper studies the effect of different QoS tools on VoIP application performance and capacity via OPNET simulation. Also, the maximum VoIP capacity which gives accepted quality will be investigated.

Algorithmic Identification of the Best WLAN Protocol and Network Architecture for Internet-Based Applications

This research developed a novel algorithm to evaluate internet-based services such as VoIP, Video Conferencing, HTTP and FTP, of different IEEE 802.11 technologies in order to identify the optimum network architecture among Basic Service Set (BSS), Extended Service Set (ESS) and the Independent Basic Service Set (IBSS). The proposed algorithm will yield the rank order of different IEEE 802.11 technologies. By selecting the optimum network architecture and technology, the best overall network performance that provides good voice, video and data quality is guaranteed. Furthermore, it meets the acceptance threshold values for the VoIP, Video Conferencing, HTTP and FTP quality metrics. This algorithm was applied to various room sizes ranging from [Formula: see text][Formula: see text]m to [Formula: see text][Formula: see text]m and the number of nodes ranged from 1 to 65. The spatial distributions considered were circular, uniform and random. The Quality of Service (QoS) metrics used were delay, jitter, throughput and packet loss.

The Influence of SIP Call Control Signalling on VoIP Quality of Experience

The Influence of SIP Call Control Signalling on VoIP Quality of Experience

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

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

Predictive model for hardware calibration to transmit real‐time applications in VoIP networks

SummaryVoice over Internet Protocol (VoIP) carries and transforms voice over the IP networks. The principles of VoIP calls are similar to traditional telephony that involves signalling, channel‐setup, digitization, and encoding of speech signal, but it transmits data over a packet‐switched network instead of circuit‐switched network. Factors which determine VoIP Quality of Service (QoS) include the choice of codec, packet loss, delay, jitter, and optimal hardware selection to handle different services. Hardware Calibration is a mechanism used for selecting an appropriate hardware for call manager to process and transmit different applications in real time. The widespread use of VoIP services in formal and informal sector produces a significant amount of data with variety of dimensions. This data can be used as leverage to analyze the system and predict various factors boosting the performance and cost effectiveness of the system. This paper proposes a predictive model that selects the best suitable hardware to handle particular offered load, which can support desired numbers of concurrent calls from wide array of processors available in the market today. This model would help the VoIP service providers in providing efficient services with QoS for different VoIP services like voice, data, video, chat, etc. This paper attempts to train and evaluate a model using various system parameters and system benchmark is predicted on an absolute scale. The results effectively demonstrate the selection of best call manager to handle offered load and hence provides QoS in overall network performance.

QoE Modeling for Voice over IP: Simplified E-model Enhancement Utilizing the Subjective MOS Prediction Model: A Case of G.729 and Thai Users

This research proposes an enhanced measurement method for VoIP quality assessment which provides an improvement to accuracy and reliability. To improve the objective measurement tool called the simplified E-model for the selected codec, G.729, it has been enhanced by utilizing a model of Mean Opinion Score (MOS), called a subjective MOS prediction model, based on native Thai users who use the Thai-tonal language. Then, the different results from the simplified E-model and subjective MOS prediction model were used to create the Bias function, before adding to the simplified E-model. Finally, it has been found that the outputs from the enhanced simplified E-model for the G.729 codec show better accuracy when compared to the original simplified E-model, specially, after the enhanced model has been evaluated with 4 test sets. The major contribution of this enhancement is that errors are reduced by 58.87% when compared to the generic simplified E-model. That means the enhanced simplified E-model as proposed in this study can provide improvement beyond the original simplified one significantly.

Fuzzy Logic based Approach for VoIP Quality Maintaining

Voice communication is an emerging technology and has great importance in our routine life. Perceptual, Voice over Internet Protocol quality is an important issue for VoIP Apps services because VoIP Apps require real-time support. Many network factors (packet loss, packet delay, and jitter) affect to VoIP quality, to achieve this objective we used an approach based on Fuzzy Logic. We configure Resource Reservation Protocol application to control Token Bucket Algorithm and the simulation experiments are carried out with Opnet. In addition, compare Token Bucket with and without Quality of Service for measure network factors. In this paper, building Fuzzy Token Bucket System consists of three variables (Bandwidth Rate, Buffer Size, and New Token) in order to improve Token Bucket Shaper output variable (New Token) by Fuzzy Stability model for Voice over IP quality maintaining.

Subjective MOS model and simplified E-model enhancement for Skype associated with packet loss effects: a case using conversation-like tests with Thai users

This paper proposes two mathematical models that can be used to estimate VoIP quality from Skype, which is one of the most popular VoIP applications. The first model is simple, it has been developed using data from the informal interview tests called Conversation-like tests, referring to packet loss of 0 %, 5 %, 10 %, …, and 30 %. The tests have been conducted with Skype using a non ITU-T’s codec called SILK via the Internet with over 180 native Thai participants, while packet loss effects were generated using a network emulation tool. The second model is called the Enhanced Simplified E-model, this has been developed by adding the Thai Bias factor into a generic Simplified E-model, which calculates by subtracting the subjective results from the computed results using the Simplified E-model formula. After obtaining the models, they were evaluated with the Test set from 36 native Thai participants (different from the other group of participants) using Mean Absolute Percentage Error technique (MAPE). It has been found that VoIP quality measurement performance of both models are classified as excellent and provide higher reliability and accuracy than the Simplified E-model. Subjective MOS model and Enhanced Simplified E-model error reduction compared to the simplified one was at about 21.9 % and 21.2 % respectively, which is the major contribution of this work.

New Signaling approach Improving the VoIP Quality of Service in Ad hoc Network

Mobile Ad Hoc NETworks (MANET) are autonomous networks composed of two or more mobile nodes with wireless communication capabilities, but they have no centralized infrastructure network. VoIP is a technology that allows you to make voice calls using an Internet connection instead of a regular (or analogy) phone line. The quality of service on the VoIP network is spread over several parameters (jitter, MOS, end to end delay, and packet loss rate) which must be taken into account and studied properly during a simulation of a network adopting this technology, which is sensitive to disturbance and dynamic topologies such as ad hoc networks. For this purpose, several studies have been conducted to improve the quality of voice conveyed via an ad hoc network known for its limited resources and constraints of a higher level. This paper presents a new approach to H.323 signalling in an ad hoc network implementing three Gatekeepers at a time, or the user calculates the distances separating the three gatekeepers to decide to connect to the appropriate Gatekeeper which is the closest geographically. This new approach is simulated on Opnet Modeler that will calculate the parameters of the obtained QoS and thus can compare them with the parameters of an H.323 signalling system based on a single Gatekeeper.

VoIP-quality of experience modeling: E-model and simplified E-model enhancement using bias factor

The E-model is a non-intrusive measurement method that many researchers have applied to the study of VoIP quality measurement. While the Simplified E-model is a modified version from the original, it can still be used as an alternative solution. Nevertheless, it has been found that the E-model and the Simplified E-model still require further improvement. Therefore, to enhance the original E-model, this paper proposes a new factor. Moreover, the Simplified E-model has also been enhanced by the same approach. Based-on the Thai environment, the new factor called Thai Bias factor, can be computed by subtracting the subjective test results using conversation tests with native Thai users from the objective test results using an E-model tool and the Simplified E-model calculation. Of course, both E-mode tests and conversation tests were conducted with the same VoIP system and test scenarios. The Enhanced E-model and the Simplified E-model using the Thai Bias factor were then evaluated by comparing the test set from other groups of native Thai users. After evaluation of the improved models, it has been found that the Enhanced E-model and the enhanced Simplified E-model can gain higher confidence. The Enhanced E-model delivers improved accuracy and reliability at approximately more than 20 % when compared to an available E-model tool, while the Enhanced Simplified E-model delivers improved performance at approximately more than 46 % when compared to Simplified E-model calculation.

Monitoring Voip Speech Quality for Chopped and Clipped Speech

Real-time monitoring of speech quality for VoIP calls is a significant challenge. This paper presents early work on a no-reference objective model for quantifying perceived speech quality in VoIP. The overall approach uses a modular design that will be able to help pinpoint the reason for degradations as well as quantifying their impact on speech quality. The model is being designed to work with narrowband and wideband signals. This initial work is focused on rating amplitude clipped or chopped speech, which are common problems in VoIP. A model sensitive to each of these degradations is presented and then tested with both synthetic and real examples of chopped and clipped speech. The results were compared with predicted MOS outputs from four objective speech quality models: ViSQOL, PESQ, POLQA and P.563. The model output showed consistent relationships between this model's clip and chop detection modules and the quality predictions from the other objective speech quality models. Further work is planned to widen the range of degradation types captured by the model, such as non-stationary background noise and speaker echo. While other components (e.g. a voice activity detector) would be necessary to deploy the model for stand-alone VoIP monitoring, the results show good potential for using the model in a realtime monitoring tool.

SEAMLESS VOIP SERVICE USING CENTRALIZED CODEC ADJUSTMENT: A FRAMEWORK IN WIRELESS NETWORK

The evolution of wireless technology leads VoIP application to be a good preferred choice among mobile users. Customers can having a VoIP service while being mobile regardless of location, terminal, and it is independent of network access. It is believed to be an alternative to the PSTN and its service is expected to be as good as PSTN. In essence, QoS plays a crucial part in order to maintain the seamless service during the VoIP session. As wireless networks are prone to channel error and VoIP application is known to be a delay-sensitive application, network traffic congestion could easily affect the voice quality of VoIP. Therefore, the codec selection within the VoIP application must be improved so that the codec can operate at various bit rates in order to cater for different traffic conditions. In this paper, we explain how centralize codec switching can help user to communicate seamlessly. We use WiFi network traffic as an example scenario.

VoIP quality measurement: subjective VoIP quality estimation model for G.711 and G.729 based on native Thai users

This paper presents a mathematical model that has been created from the subjective MOS, instead of modifying or improving the existing objective measurement methods (e.g., E-model) for VoIP quality measurement. The proposed model of VoIP quality measurement method is based on native Thai users who communicate to each other using Thai language, which is a tonal language, unlike English and most western languages. The data have been gathered using conversation-opinion tests with 400 and 354 native Thai subjects for two popular codecs, G.711 and G.729, respectively, referring to effects from two major network factors, packet loss and packet delay. This model is called the Thai subjective VoIP quality evaluation model (ThaiVQE). It has been evaluated using two test sets of subjective MOS, from 50 native Thai subjects for G.711 and 64 native Thai subjects for G.729, then the results have been compared with the E-model results. Based on native Thai users, the evaluation result surprisingly shows that ThaiVQE can contribute better accuracy and reliability than the standard E-model with error reduction of over 13 % for G.711 and 28 % for G.729. Therefore, this is an example study for other countries that have their own languages and cultures to create their subjective MOS model.

A QoS enabled multimedia WiFi access point

SummaryThe Internet has evolved to a stage where the number of Internet enabled devices exceeds the global population. With much of the required connectivity over wireless, contention for bandwidth is an important issue that has to be addressed. In this context, applications that have certain quality of service (QoS) requirements must be protected. WiFi‐enabled mobile devices such as smartphones and tablets are commonly used by both personal and business users for voice over IP (VoIP) communications, while also supporting conventional data applications such as email, file transfer and web access. Previous research has shown that time synchronized endpoints can provide better QoS by calculating accurate mouth‐to‐ear delays, and using this information to better inform buffer playout strategies. The IEEE 802.11e protocol extends 802.11 by providing different traffic priorities based on traffic type. However, it cannot distinguish between traffic streams within the same category. In this paper, we present an access point‐centred mechanism for further distinguishing between streams within 802.11e categories. The mechanism predicts the VoIP quality for multiple sessions using the ITU‐T E‐Model. Delay values are determined from RTCP packet timing information. We provide implementation details on a real world proof‐of‐concept and present results that correlate well with previous NS‐3 based simulations. In addition, we address scalability issues for large scale deployments. Copyright © 2015 John Wiley & Sons, Ltd.

Audio Codecs Impact on Quality of VoIP Based on IEEE802.16e Considering Mobile IP Handover

A simulation model using OPNET tool is introduced for testing audio codecs impact on quality of VoIP based on IEEE802.16e with taking into account handover performance. Different parameters that indicate the quality of VoIP such as Throughput, MOS, End to End delay and traffic send and received. According to our simulation the acceptable MOS value is recorded for codecs G.711 and GSM-FR.