Voice Over Internet Protocol Quality Research Articles

Voice over Internet Protocol over Wireless Local Area Network: A Review

The use of Voice over Wireless Local Area Network is seeing a meteoric rise in popularity as a result of its simplicity, non-intrusiveness, and cheap cost of implementation, as well as its low cost of maintenance, universal coverage, and fundamental roaming capabilities. Nevertheless, deploying Voice over Internet Protocol (VoIP) over Wireless Local Area Network (WLAN) is a challenging task for many network managers, architects, planners, designers, and engineers. Because of this, there is a need for a guideline to design, model, and simulate the network before it is deployed. In this work, a variety of models, including mathematical, theoretical, statistical, and graphical models, that are used to measure the quality and features of VoIP are discussed.

Enhanced Routing Algorithm Based on Reinforcement Machine Learning-A Case of VoIP Service.

The routing algorithm is one of the main factors that directly impact on network performance. However, conventional routing algorithms do not consider the network data history, for instances, overloaded paths or equipment faults. It is expected that routing algorithms based on machine learning present advantages using that network data. Nevertheless, in a routing algorithm based on reinforcement learning (RL) technique, additional control message headers could be required. In this context, this research presents an enhanced routing protocol based on RL, named e-RLRP, in which the overhead is reduced. Specifically, a dynamic adjustment in the Hello message interval is implemented to compensate the overhead generated by the use of RL. Different network scenarios with variable number of nodes, routes, traffic flows and degree of mobility are implemented, in which network parameters, such as packet loss, delay, throughput and overhead are obtained. Additionally, a Voice-over-IP (VoIP) communication scenario is implemented, in which the E-model algorithm is used to predict the communication quality. For performance comparison, the OLSR, BATMAN and RLRP protocols are used. Experimental results show that the e-RLRP reduces network overhead compared to RLRP, and overcomes in most cases all of these protocols, considering both network parameters and VoIP quality.

Studying the Impact of SIP Message Differentiation on the Quality of VoIP Session Control Procedures

Voice over Internet Protocol (VoIP) is characterized by the growth of session control signalling, which is based on the Session Initiation Protocol (SIP). The growing amount of SIP signalling may cause an overload of VoIP session control infrastructure and quality of service degradation. To reduce the effect of overload conditions, we studied the impact of SIP message differentiation on the quality of VoIP session control procedures. We proposed an algorithm for SIP message prioritization and tested it with Kamailio SIP server under different conditions. Summarizing our research findings, the proposed algorithm improves the performance metrics related to VoIP session control procedures.

Improved voice quality with the combination of transport layer & audio codec for wireless devices

Improving voice quality over wireless communication becomes a demanding feature for social media apps like facebook, whatsapp and other communication channels. Voice-over-internet protocol (VoIP) helps us to make quick telephone calls over the internet. It includes various mechanism which are signaling, controlling and transport layer. Over wireless links, packet loss and high transmission delay damage voice quality. Here VoIP quality will be measured by three main elements which are signaling protocol, audio codec and transport layer. To improve the overall voice quality, we need to combine these three elements properly to get the best score. Otherwise perceptual speech quality will not be the right tool to measure the voice quality. Here we will use Mean Opinion Score (MOS) for calculated jitter values and end to end delay. At the end, best combination of audio codec signaling protocol produced the quality speech.

Jitter Buffer Control Algorithm and Simulation Based on Network Traffic Prediction

The jitter buffer size is one of the important factors that affect the voice quality of Voice over Internet Protocol (VoIP). In order to reduce buffer delay and packet loss and improve the speech quality of VoIP, an adaptive jitter buffer control algorithm based on network traffic prediction is proposed. Among them, the particle swarm optimization algorithm is used to optimize the weight and threshold of Elman neural network so that it avoids falling into local minimum and improves the accuracy of network traffic prediction. According to the change law of network traffic, the jitter buffer control algorithm under the autoregressive model is proposed. Through the improved stochastic midpoint placement algorithm, a network business traffic prediction model with sudden and self-similarity is established. Then the buffer size is set according to the predicted value of network traffic, and it is continuously improved in use to improve the accuracy of the buffer setting. The simulation results show that the MOS value of the algorithm is high, which improves the voice quality of the network telephone.

Performance analysis of VoIP under the effect of interference and during conference call in WLAN network using OPNET modeller

Circuit switched network or packet switched networks are used for both visual and vocal communication. Public switched telephone network (PSTN) is not an affordable option when comparing with existing packet switched network. Voice over internet protocol (VoIP) has become a preferable alternative due to its reduced cost and flexibility compared to voice over LTE (VoLTE) where carriers are still building out their 4G networks. However, despite its reduced cost, it has to face so many challenges which affect its successful deployment. The reason is that the quality of VoIP is mainly affected by jitter, delay, packet loss and various other parameters. This research was carried out to evaluate the quality of voice in VoIP experimentally, under the effect of interference and during conference calls. The simulations were carried out using Riverbed Modeler academic edition 17.5. The results of the analysis and the performance evaluation are presented in this paper. This work can guide researchers and designers to design a network for VoIP services and its deployment based on their requirements.

An optimized hardware calibration technique for transmission of real-time applications in VoIP network

Voice over Internet Protocol (VoIP) uses packet switching to transmit voice, video, data and chat services. VoIP applications depend upon various speech codecs which vary in speech quality, bandwidth and computational requirements. The key challenges for improving the VoIP Quality of Service (QoS) and Quality of Experience (QoE) is to use best codec by considering network traffic and selecting the optimal hardware. Hardware Calibration is a mechanism to aid service providers by selecting a suitable hardware for Call Manager to transmit different real-time based applications. The paper focuses on proposing a computational model which suggests the most suitable hardware for a specific load handling requirement. The selection of processor is done by taking into consideration the effect of different codecs when it takes calls per second as input and gives output as whichever hardware is recommended for Call Manager to handle different types of applications like Peer-To-Peer (P2P) application, Back-To-Back (BTB) application, enabling voice logging, etc. The emulations demonstrate that the proposed model is suitable for selection of hardware which provides better QoS and QoE for the transmission VoIP based real-time applications. Further, a multi-criteria decision-making method based on Analytic Hierarchy Process (AHP) is also proposed to decide which hardware can be chosen as Call Manager. The results obtained from AHP are used to validate the outcome of the proposed computational model. The computational model is applicable only when the tests are run on particular systems and it suggests the best hardware among them for the particular offered load. However, to suggest the best configuration corresponding to the input load from the possible configurations available in the market, benchmarking is performed. A common scale model for hardware benchmark estimation particularly has been proposed for different scenarios.

An efficient and secure VoIP communication system with chaotic mapping and message digest

As one of the most popular real-time services on the Internet, Voice over internet protocol (VoIP) has attracted researchers' attention in the information security field for its characters of real-time and high flow. To protect data security, a new covert VoIP communications system was proposed in this study to realize secure communications by hiding secret information in VoIP streams. In the proposed algorithm, secret data were divided into blocks after being encrypted with a block cipher, and then each block of secret data was embedded into VoIP streaming packets randomly using a chaotic mapping. The symmetric key was distributed through an efficient and secure channel, and the message digest was implemented to protect the integrity of secret data. The experimental data were collected by comparing audio data between the sender and the receiver. The experimental results indicated that data embedding had little impact on the quality of speech. Besides, statistical analysis could not detect the secret data embedded in VoIP streams using the block cipher and random numbers generated from chaotic mapping. The proposed covert VoIP communications system not only achieved high quality of VoIP and prevented detection of statistical analysis, but also provided integrity for secret data.

Implementation of Hotspot Network for Internal Campus Communications Utilizing Smartphone and Free Software

VoIP (voice over internet protocol) telephone communication using a data network. There is a change in switching from circuit switching technology into packet switching. Phone exchange can now use the Personal Computer equipped VoIP applications. Even the development of mobile phone technology to make VoIP communication can be performed utilizing the Smartphone. VoIP applications commonly use such as WhatsApp, Line etc. However, this application will cut the user data packets and often the quality is not satisfactory due to limited bandwidth and location of the remote server. During this time, Udayana University campus at several locations has been equipped with Hotspot network is mostly used to connect to the Internet. Hotspot network can also be used for voice communications with VoIP technology by adding a VoIP server. This concept not raises communication costs and should produce sound quality will be better because of the close location of the server. Because that researchers need to develop a model of telephone communication network utilizing Smartphone hotspot and students to be able to communicate in a campus environment. Method of this research is to develop a network utilizing hotspots and VoIP telephone exchange using the mini PC installed software FreePBX. On the side of the Smartphone using the free soft phone application and for aircraft used FXS analog phone as a codec. Tests will be performed for the communication between Smartphone and Smartphone to FXS terms of QOS and MOS produced. The results obtained, if the latency and packet loss have a value corresponding to the Real Time Protocol (RTP), the obtained MOS appropriate for the codec used while with the same codec if the value of packet loss and latency results are high then MOS obtained be small or less good quality. So the quality of VoIP is highly dependent on the quality of the signal obtained hotspot. In general, VoIP communication using a Smartphone connected to the server on the network hotspot mini pc can be used as a voice communication on campus.

Security of VoIP traffic over low or limited bandwidth networks

AbstractThe early days of voice over IP (VoIP) adoption were characterized by a lack of concern and awareness about security issues related to its use. Indeed, service providers and users were mostly preoccupied with issues related to its quality, functionality, and cost. Now that VoIP is a mainstream communication technology, security has become a major issue. This paper investigates the major security threats for VoIP communications and proposes a multipath approach solution, especially targeted for low bandwidth networks. Results show that security has an effect on VoIP quality especially for a large distance between communicating nodes and packet size. Results also show that our proposed multipath solution reduces significantly packet losses and performs better than single routing techniques in networks with low bandwidth capacities. Copyright © 2017 John Wiley & Sons, Ltd.

Packet dispersion strategy evaluation from the perspective of packet loss pattern and VoIP quality

The appearance of burst packet losses and its devastating effect on Voice over IP (VoIP) service have imposed a requirement for the implementation of loss recovery mechanisms to address VoIP quality during periods when high packet loss is exhibited. Existing loss recovery mechanisms are dependent on end point capabilities, whereas Quality of service (QoS) routing protocols suffer from complexity and scalability issues. In this paper, we examine packet dispersion?s ability to address burst losses and provide a computational model, which is verified using real network testing. A study has been carried out to investigate the effect of different packet dispersion strategies on burst losses, which clearly shows dispersion?s qualitative superiority over single path routing. Furthermore, an analytical approach is proposed resulting in quality estimation obtained by individual strategies. Practical evaluation has shown that each strategy copes differently with various burst scenarios in order to maximize VoIP quality.

A New Distributed Application and Network Layer Protocol for VoIP in Mobile Ad Hoc Networks

In this work a new protocol for Voice over IP (VoIP) transmissions in wireless ad-hoc networks is proposed. Distributed architecture is necessary when dealing with dynamic environments, such as ports or battlefields, where creating infrastructures becomes expensive or impossible. Mobile ad-hoc networks (MANETs) are based on a peer-to-peer approach and each node participates in the organization of the whole network. VoIP over MANETs is a challenging issue due to the intrinsic distributed nature of the existing peer-to-peer paradigm. This paper proposes a new protocol, capable of ensuring a quality of service (QoS) level for VoIP calls over a MANET and to manage a large number of calls in the system. Novel metric and utility functions are proposed to perform the best path selection from source to destination nodes, respecting the QoS parameters for VoIP quality. In particular, an objective metric such as R-factor is considered, and a flexibility index is defined in order to maximize the number of acceptable VoIP calls. Performance evaluation shows that the proposed approach led to better network management in terms of admitted calls and respected QoS constraints.

Burst packet loss and enhanced packet loss‐based quality model for mobile voice‐over Internet protocol applications

Voice over Internet protocol (VoIP) quality monitoring platforms need to be as simple as possible, especially at low capacity end-points, such as smart phones. The existing simple packet loss-based quality model, with only the input metric of the packet loss parameter, needs to be enhanced to increase the accuracy of quality prediction while ensuring that implementation is simple. Several burstiness parameters for describing burst loss characteristics in VoIP packet streams has been presented. The authors investigate how closely each one is correlated to perceptual quality. Then, from the correlation results, two dominant parameters are chosen that will be used as additional input metrics to increase the accuracy of quality prediction. A packet loss-based and enhanced quality prediction model is also proposed for VoIP services over the Internet. This can replace the speech quality estimation model in ITU G.1070, which aims to implement VoIP quality monitoring platforms at low capacity end-points. Thus, mobile end-points, such as smart phones, can embed the proposed quality prediction model to measure VoIP quality on a real-time basis.

A New QoS Management Scheme for VoIP Application over Wireless Ad Hoc Networks

Nowadays, mobile ad hoc networks (MANETs) have to support new applications including VoIP (voice over IP) that impose stringent QoS (quality of service) constraints and requirements. However, VoIP applications make a very inefficient use of the MANET resources. Our work represents a first step toward improving aspects at the network layer by addressing issues from the standpoint of adaptation, claiming that effective adaptation of routing parameters can enhance VoIP quality. The most important contribution is the adaptive OLSR-VA algorithm which provides an integrated environment where VoIP activity is constantly detected and routing parameters are adapted in order to meet the application requirements. To investigate the performance advantage achieved by such algorithm, a number of realistic simulations (MANET scenarios) are performed under different conditions. The most important observation is that performance is satisfactory in terms of the perceived voice quality.

Selection of VoIP CODECs for Different Networks based on QoS Analysis

Voice over Internet Protocol (VoIP) has been an interesting topic of research in the last decade. The engrossing increase in the use of VoIP services is resulting in the enormous growth of broadband network. The main objective of this paper is the selection of an appropriate voice compression and decompression (CODEC) schemes depending on the Quality of Service (QoS) of VoIP in different networks. Wired, Wireless Local Area Network (WLAN), Worldwide Interoperability for Microwave Access (WiMAX) and Universal Mobile Telecommunication System (UMTS) networks were implemented in OPNET Modeler. The quality is compared using different QoS parameters like end-to-end delay, MOS, throughput and jitter. The VoIP codecs used in the measurements of QoS are: GSM-FR, G.711, G.723.1 and G.729A. Simulations showed that G.711 and GSMFR are the best schemes that provide high quality of voice in Wireless Local Area Network (WLAN) communications. In WiMAX, G.729A gives the best quality of VoIP while in UMTS, GSMFR gives overall best results with respect to all the parameters. Wired model gives the best result irrespective of the codec being used. G.723.1 can be used in WiMAX and UMTS along with the wired network depending on conditions. The results analyzed and the performance evaluated will give network operators an opportunity to select the codec for better services of VoIP for customer satisfaction.

IVoIP: an intelligent bandwidth management scheme for VoIP in WLANs

Voice over Internet Protocol (VoIP) has been widely used by many mobile consumer devices in IEEE 802.11 wireless local area networks (WLAN) due to its low cost and convenience. However, delays of all VoIP flows dramatically increase when network capacity is approached. Additionally, unfair traffic distribution between downlink and uplink flows in WLANs impacts the perceived VoIP quality. This paper proposes an intelligent bandwidth management scheme for VoIP services (iVoIP) that improves bandwidth utilization and provides fair downlink---uplink channel access. iVoIP is a cross-layer solution which includes two components: (1) iVoIP-Admission Control, which protects the quality of existing flows and increases the utilization of wireless network resources; (2) iVoIP-Fairness scheme, which balances the channel access opportunity between access point (AP) and wireless stations. iVoIP-Admission Control limits the number of VoIP flows based on an estimation of VoIP capacity. iVoIP-Fairness implements a contention window adaptation scheme at AP which uses stereotypes and considers several major quality of service parameters to balance the network access of downlink and uplink flows, respectively. Extensive simulations and real tests have been performed, demonstrating that iVoIP has both very good VoIP capacity estimation and admission control results. Additionally, iVoIP improves the downlink/uplink fairness level in terms of throughput, delay, loss, and VoIP quality.

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.

Quality of VoIP in WiMAX and UMTS

The deployment of the Next Generation Wireless Networks (NGWNs) has posed great challenges due to the demanding requirements to integrate different radio access technologies and to provide high Quality of Service (QoS) for applications such as Voice over IP (VoIP) traffic. To meet the demand of providing high-quality of VoIP at anytime and from anywhere, it is imperative to design suitable models to map QoS between different radio networks. In this paper we investigate the QoS performance of WiMAX and UMTS for supporting VoIP applications since UMTS and WiMAX are emerging as the promising technologies for building the NGWNs. We designed simulation modules in OPNET for both WiMAX andUMTS, and carried out extensive simulations to evaluate and analyze several important performance metrics such as Mean Opinion Score (MOS), end-to-end delay, jitter and packet delay variation. Simulation results show that WiMAX outscores UMTS with a sufficient margin, and provides better support to VoIP applications compared with UMTS.

Performance Analysis of Various Codecs using RSVP on VoIP Quality of Service over Variable Bandwidth

With the rapid growth of the Internet, customers are demanding multimedia applications such as telephony and video on demand, to be available on the internet. Voice over Internet Protocol (VoIP) is a rapidly emerging technology for voice communication that uses the ubiquity of IP-based networks. The greatest challenge is the provisioning of Quality of Service (QoS) over limited bandwidth. Multimedia applications need a lot more bandwidth and have different QoS requirements than the applications that were used in early years of the internet. The purpose of this paper is to check the performance of VoIP application under different Codec’s such as G.711, G.729 and G.723.1 over variable bandwidth. This paper proposes RSVP protocol for providing end to end QoS for real time applications over diverse networks. The performance is checked under RSVP and Non RSVP. Packet end to end delay and packet delay variation are evaluated through simulation. The result shows that under any bandwidth G.723.1 gives better result as compared to G.711 and G.729. Simulation is done using OPNET IT Guru Academic edition.

Mobile WiMAX field trial test through multimedia performance evaluation

IEEE 802.16e is a mobile version of Worldwide Interoperability for Microwave Access (WiMAX) that plays an important role in the evolution towards 4G. In this study, we focus on multimedia performance measurement for the purpose of a more realistic mobile WiMAX network test. This study aims to make a contribution in better understanding the mobile WiMAX performance for multimedia applications. For that purpose, we employ Voice over Internet Protocol (VoIP) and video streaming to test the network performance, where two distinct evaluation systems are used, professional and user-friendly. Our test results show that the mobile WiMAX network can support well the bandwidth-intense and delay-sensitive multimedia application. We find that the VoIP quality at the cell center is perfect, where the value of Perceptual Evaluation of Speech Quality exceeds 4. At the cell edge, the quality is degraded but still adequate. We also observe that the downlink of mobile WiMAX network can support video streaming up to 4 Mbps with the Mean Opinion Score (MOS) value of 4.5. On the uplink, the bitrate of 1 Mbps is supported with MOS 4.5 at the cell center and with MOS 3.2 at the cell edge, respectively. Our experiments further indicate that a smooth playback of YouTube 480P video is consistently provided. Finally, the handover case has very limited impact to the overall quality degradation of both VoIP and video streaming.